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24.
Live Instrument Reference

Live comes with a selection of custom-designed, built-in instruments. The Working with Instruments and Effects chapter (see Chapter 17) explains the basics of using instruments in Live.

24.1
Analog

The Analog Instrument.

Analog is a virtual analog synthesizer, created in collaboration with Applied Acoustics Systems. With this instrument, we have not attempted to emulate a specific vintage analog synthesizer but rather to combine different features of legendary vintage synthesizers into a modern instrument. Analog generates sound by simulating the different components of the synthesizer through physical modeling. This technology uses the laws of physics to reproduce how an object or system produces sound. In the case of Analog, mathematical equations describing how analog circuits function are solved in real time. Analog uses no sampling or wavetables; the sound is simply calculated in real time by the CPU according to the values of each parameter. This sound synthesis method ensures unmatched sound quality, realism, warmth and playing dynamics.

The full version of Analog is not included with the standard version of Live, but is a special feature available for purchase separately.

24.1.1
Architecture and Interface

Analog’s signal flow is shown in the figure below:

Diagram of Analog’s Signal Flow.

The primary sound sources of the synthesizer are two oscillators and a noise generator. These sources can be independently routed to two different multi-mode filters, which are each connected to an amplifier. Furthermore, the signal flow can be run through the filters in series or in parallel.

Analog also features two low-frequency oscillators (LFOs) which can modulate the oscillators, filters and amplifiers. Additionally, each filter and amplifier has its own envelope generator.

The Analog interface consists of two parts: the
display
surrounded on all sides by the
shell
. The shell contains the most important controls for a given section while the display updates to show parameter visualizations and additional controls for the selected section. In addition to the synthesis modules, there is a Global section that contains general performance parameters such as instrument volume, vibrato and polyphony.

24.1.2
Oscillators

Display and Shell Parameters for the two Oscillators.

Analog’s two oscillators use physical modelling to capture the character of vintage hardware oscillators. Because they use modelling instead of wavetables, they avoid aliasing.

Each oscillator can be turned on or off independently via the switch labelled Osc 1 or Osc 2 in the shell, and the oscillator’s output level is adjusted by the slider to the right of this activator.

The F1/F2 slider controls the balance of the oscillator’s output to each of the two filters. When the slider is at the center position, equal amounts of signal will be sent to both filters. When set all the way to the top or bottom, signal will only be sent to Filter 1 or Filter 2 respectively.

The Shape chooser selects the oscillator’s waveform. The choices are sine, sawtooth, rectangular and white noise. When rectangular is selected, the Pulse Width parameter is enabled in the display, which allows you to change the pulse width of the waveform. Low Width values result in a very narrow waveform, which tends to sound tinny or “pinched.“ At 100%, the waveform is a perfect square, resulting in only odd harmonics. The pulse width can also be modulated by an LFO, via the slider next to Width. Note that this parameter is only enabled when the corresponding LFO is enabled.

The Octave, Semi and Detune knobs in the shell function as coarse and fine tuners. Octave transposes the oscillator by octaves, while Semi transposes up or down in semitone increments. The Detune knob adjusts in increments of one cent (up to a maximum of three semitones (300 cents) up or down).

Oscillator pitch can be modulated according to the settings of the Pitch Mod and Pitch Env parameters in the display. The LFO slider sets the amount that the LFO modulates pitch. Again, this parameter is only enabled if the LFO is on. The Key slider controls how much the oscillator tuning is adjusted by changes in MIDI note pitch. The default value of 100% means that the oscillator will conform to a conventional equal tempered scale. Higher or lower values change the amount of space between the notes on the keyboard. At 0%, the oscillator is not modulated by note pitch at all. To get a sense of how this works, try leaving one of the oscillators at 100% and setting the other’s Key scaling to something just slightly different. Then play scales near middle C. Since C3 will always trigger the same frequency regardless of the Key value, the oscillators will get farther out of tune with each other the farther away from C3 you play.

The Pitch Env settings apply a ramp that modulates the oscillator’s pitch over time. Initial sets the starting pitch of the oscillator while Time adjusts how long it will take for the pitch to glide to its final value. You can adjust both parameters via the sliders or by adjusting the breakpoints in the envelope display.

The Sub/Sync parameters in the display allow you to apply either a sub-oscillator or a hard synchronization mode. When the Mode chooser is set to Sub, the Level slider sets the output level of an additional oscillator, tuned an octave below the main oscillator. The sub-oscillator produces a square wave when the main oscillator’s Shape control is set to rectangle or sawtooth and a sine wave when the main oscillator is set to sine. Note that the sub-oscillator is disabled when the main oscillator’s Shape is set to white noise.

When the Mode chooser is set to Sync, the oscillator’s waveform is restarted by an internal oscillator whose frequency is set by the Ratio slider. At 0%, the frequency of the internal oscillator and the audible oscillator match, so sync has no effect. As you increase the Ratio, the internal oscillator’s rate increases, which changes the harmonic content of the audible oscillator. For maximum analog nastiness, try mapping a modulation wheel or other MIDI controller to the Sync ratio.

24.1.3
Noise Generator

Analog’s Noise Generator.

The Noise generator produces white noise and includes its own -6db/octave low-pass filter. The generator can be turned on or off via the Noise switch in the shell. Its output level is adjusted by the slider to the right of this activator.

The F1/F2 slider controls the balance of the noise generator’s output to each of the two filters. When the slider is at the center position, equal amounts of signal will be sent to both filters. When set all the way to the top or bottom, signal will only be sent to Filter 1 or Filter 2 respectively.

The Color knob sets the frequency of the internal low-pass filter. Higher values result in more high-frequency content.

Note that Noise has only shell parameters, so adjusting them does not change what is shown in the display.

24.1.4
Filters

Display and Shell Parameters for the two Filters.

Analog’s two multi-mode filters come equipped with a flexible routing architecture, multiple saturation options and a variety of modulation possibilities. As with the oscillators, all parameters can be set independently for each filter.

The Fil 1 and Fil 2 switches in the shell toggle the respective filter on and off. The chooser next to the filter activator selects the filter type from a selection of 2nd and 4th order low-pass, band-pass, notch, high-pass and formant filters.

The resonance frequency of the filter is adjusted with the Freq knob in the shell, while the amount of resonance is adjusted with the Reso control. When a formant filter is chosen in the chooser, the Reso control cycles between vowel sounds.

Below each mode chooser is an additional control which differs between the two filters. In Filter 1, the To F2 slider allows you to adjust the amount of Filter 1’s output that will be sent to Filter 2. The Slave switch below Filter 2’s mode chooser causes this filter’s cutoff frequency to follow the cutoff of Filter 1. If this is enabled, Filter 2’s cutoff knob controls the amount of offset between the two cutoff amounts. If any of Analog’s modulation sources are controlling Filter 1’s cutoff, Filter 2 will also be affected by them when Slave is enabled.

In addition to the envelope controls (see 24.1.6), the displays for the filters contain various modulation parameters and the Drive chooser. Cutoff frequency and resonance can be independently modulated by LFO, note pitch and filter envelope via the sliders in the Freq Mod and Res Mod sections respectively. Positive modulation values will increase the cutoff or resonance amounts, while negative values will lower them.

The Drive chooser in the display selects the type of saturation applied to the filter output. The three Sym options apply symmetrical distortion, which means that the saturation behavior is the same for positive and negative values. The Asym modes result in asymmetrical saturation. For both mode types, higher numbers result in more distortion. Drive can be switched off entirely by selecting Off in the chooser. Experiment with the various options to get a sense of how they affect incoming signals.

24.1.5
Amplifiers

Display and Shell Parameters for the two Amplifiers.

After the filters, the signal is routed to an amplifier which further shapes the sound with an amplitude envelope and panning. All parameters can be set independently for each amplifier.

The Amp 1 and Amp 2 switches in the shell toggle the respective amplifier on and off, while the output level is controlled by the Level knob. The Pan knob sets the position of the amplifier’s output in the stereo field.

In addition to the envelope controls, the displays for the amplifiers contain various modulation parameters. The Pan and Level amounts can be independently modulated by LFO, note pitch and amp envelope via the sliders in the Pan Mod and Level Mod sections respectively. Note that, when using note pitch as the modulation source for Level, middle C will always sound the same regardless of the modulation amount. Positive values will cause the level to increase for higher notes.

24.1.6
Envelopes

Analog’s Envelope Parameters.

In addition to the pitch envelopes in the oscillator sections, Analog is equipped with independent envelopes for each filter and amplifier. All four of these envelopes have identical controls, which are housed entirely within the display. Each envelope is a standard ADSR (attack, decay, sustain, release) design and features velocity modulation and looping capabilities.

The attack time is set with the Attack slider. This time can also be modulated by velocity via the Att

The time it takes for the envelope to reach the sustain level after the attack phase is set by the Decay slider.

The Sustain slider sets the level at which the envelope will remain from the end of the decay phase to the release of the key. When this knob is turned all the way to the left, there is no sustain phase. With it turned all the way to the right, there is no decay phase.

The overall envelope level can be additionally modulated by velocity via the Env

The S.Time slider can cause the Sustain level to decrease even if a key remains depressed. Lower values cause the Sustain level to decrease more quickly.

Finally, the release time is set with the Release knob. This is the time it takes for the envelope to reach zero after the key is released.

The Slope switches toggle the shape of the envelope segments between linear and exponential. This change is also represented in the envelope visualization.

Normally, each new note triggers its own envelope from the beginning of the attack phase. With Legato enabled, a new note that is played while another note is already depressed will use the first note’s envelope, at its current position.

Enabling the Free switch causes the envelope to bypass its sustain phase and move directly from the decay phase to the release phase. This behavior is sometimes called “trigger“ mode because it produces notes of equal duration, regardless of how long the key is depressed. Free mode is ideal for percussive sounds.

The Loop chooser offers several options for repeating certain segments of the envelope while a key is depressed. When Off is selected, the envelope plays once through all of its segments without looping.

With AD-R selected, the envelope begins with the attack and decay phases as usual, but rather than maintaining the sustain level, the attack and decay phases will repeat until the note is released, at which point the release phase occurs. ADR-R mode is similar, but also includes the release phase in the loop for as long as the key is held.

Note that in both AD-R and ADR-R modes, enabling Free will cause notes to behave as if they’re permanently depressed.

ADS-R mode plays the envelope without looping, but plays the attack and release phases once more when the key is released. With short attack and release times, this mode can simulate instruments with audible dampers.

24.1.7
LFOs

Display and Shell Parameters for the two LFOs.

Analog’s two LFOs can be used as modulation sources for the oscillators, filters and amplifiers. As with the other sections, each LFO has independent parameters.

The LFO 1 and LFO 2 switches in the shell toggle the respective LFO on and off, while the Rate knob sets the LFO’s speed. The switch next to this knob toggles the Rate between frequency in Hertz and tempo-synced beat divisions.

The Wave chooser in the display selects the waveform for the LFO. The choices are sine, triangle, rectangle and two types of noise. The first noise type steps between random values while the second uses smooth ramps. With Tri or Rect selected, the Width slider allows you to adjust the pulse width of the waveform. With Tri selected, low Width values shift the waveform towards an upwards sawtooth, while higher values result in a downward saw. At 50%, the waveform is a perfect triangle. The behavior is similar with the Rect setting. At 50%, the waveform is a perfect square wave, while lower and higher values result in negative or positive pulses, respectively. Note that Width is disabled when the LFO’s waveform is set to sine or the noise modes.

The Delay slider sets how long it will take for the LFO to start after the note begins, while Attack sets how long it takes the LFO to reach its full amplitude.

With Retrig enabled, the LFO restarts at the same position in its phase each time a note is triggered. The Offset slider adjusts the phase of the LFO’s waveform.

24.1.8
Global Parameters

Display and Shell Parameters for the Global Options.

The Global shell and display parameters adjust how Analog responds to MIDI data, as well as controls for performance parameters such as vibrato and glide.

The Volume control in the shell adjusts the overall output of the instrument. This is the instrument’s master level, and can boost or attenuate the output of the amplifier sections.

The Vib switch turns the vibrato effect on or off, while the percentage slider next to it adjusts the amplitude of the vibrato. Analog’s vibrato effect is essentially an additional LFO, but is hardwired to the pitch of both oscillators. The Rate slider sets the speed of the vibrato.

Turning on the vibrato effect enables the four additional Vibrato parameters in the display. The Delay slider sets how long it will take for the vibrato to start after the note begins, while Attack sets how long it takes for the vibrato to reach full intensity.

The Error slider adds a certain amount of random deviation to the Rate, Amount, Delay and Attack parameters for the vibrato applied to each polyphonic voice.

The Amt

The Uni switch in the shell turns on the unison effect, which stacks multiple voices for each note played. The Detune slider next to this switch adjusts the amount of tuning variation applied to each stacked voice.

Turning on the unison effect enables the two additional Unison parameters in the display. The Voices chooser selects between two or four stacked voices, while the Delay slider increases the lag time before each stacked voice is activated.

The Gli switch turns the glide effect on or off. This is used to make the pitch slide between notes rather than changing immediately. With Legato enabled, the sliding will only occur if the second note is played before the first note is released. The Time slider sets the overall speed of the slide.

Turning on the glide effect enables an additional Glide Mode chooser in the display. Selecting Const causes the glide time to be constant regardless of interval. Choosing Prop (proportional) causes the glide time to be proportional to the interval between the notes. Large intervals will glide slower than small intervals.

The Keyboard section in the display contains all of Analog’s polyphony and tuning parameters. The Voices chooser sets the available polyphony, while Priority determines which notes will be cut off when the maximum polyphony is exceeded. When Priority is set to High, new notes that are higher than currently sustained notes will have priority, and notes will be cut off starting from the lowest pitch. Low is the opposite. A Priority setting of Last gives priority to the most recently played notes, cutting off the oldest notes as necessary.

The Octave, Semi and Tuning controls function as coarse and fine tuners. Octave transposes the entire instrument by octaves, while Semi transposes up or down in semitone increments. The Tuning slider adjusts in increments of one cent (up to a maximum of 50 cents up or down).

PB Range sets the range in semitones of pitch bend modulation.

Stretch simulates a technique known as stretch tuning, which is a common tuning modification made to electric and acoustic pianos. At 0%, Analog will play in equal temperament, which means that two notes are an octave apart when the upper note’s fundamental pitch is exactly twice the lower note’s. Increasing the Stretch amount raises the pitch of upper notes while lowering the pitch of lower ones. The result is a more brilliant sound. Negative values simulate “negative“ stretch tuning; upper notes become flatter while lower notes become sharper.

The Error slider increases the amount of random tuning error applied to each note.

The four Quick Routing buttons in the left side of the display provide an easy way to quickly set up common parameter routings. The upper left option configures a parallel routing structure, with each oscillator feeding its own filter and amplifier exclusively. The upper right button is similar, but the oscillators each split their output evenly between the two filters. The bottom left option feeds both oscillators into Filter 1 and Amp 1, disabling Filter 2 and Amp 2 entirely. Finally, the bottom right option configures a serial routing structure, with both oscillators feeding Filter 1, which is then fed exclusively to Filter 2 and Amp 2.

Note that the Quick Routing options do not affect any changes you may have made to the oscillator level, tuning or waveform parameters — they only adjust the routing of the oscillators to the filters and subsequent amplifiers.

24.2
Collision

The Collision Instrument.

Collision is a synthesizer that simulates the characteristics of mallet percussion instruments. Created in collaboration with Applied Acoustics Systems, Collision uses physical modeling technology to model the various sound generating and resonant components of real (or imagined) objects.

The full version of Collision is not included with the standard version of Live, but is bundled with the Corpus effect (see 22.8) as a special feature available for purchase separately.

24.2.1
Architecture and Interface

Collision’s sound is produced by a pair of oscillators called
Mallet
and
Noise
, which feed a pair of independent (or linked) stereo
resonators
. While the oscillators produce the initial component of the sound, it is the resonator parameters that have the greatest impact on the sound’s character.

Collision’s interface is divided into tabs, which are further divided into sections. The
Excitator
tab contains the controls for the Mallet and Noise oscillators. The
Resonator
tabs contain the parameters for the independent resonator sections, while the
Link
tab allows you to adjust both resonators simultaneously.

The
LFO
tab contains two independent low-frequency oscillators (LFOs), which can each modulate multiple parameters. Similarly, the
MIDI
tab allows for MIDI pitch bend, modulation wheel and aftertouch messages to be routed to multiple destinations.

To the right of the resonators is a section of
global
parameters, including overall output volume, polyphony and resonator routing options.

In addition to serving as organizational aids, Collision’s tabs contain LEDs that light up to indicate that their contained sections are active. Disabling unused sections can save CPU.

24.2.2
Excitator Tab

The Excitator tab contains parameters for the Mallet and Noise sections. These model the behavior of a mallet striking a surface, and provide Collision’s fundamental sound. These section’s parameters only control the initial impulse, which is a much smaller component of Collision’s overall sound than the resonators.

Note that if both the Mallet and Noise sections are turned off, Collision will not make sound.

The Mallet Section

Collision’s Mallet Section.

The Mallet section simulates the impact of a mallet against a surface. The parameters adjust the physical properties of the mallet itself.

Volume controls the overall output level of the mallet section, while the Noise knob sets the amount of impact noise that is included in each mallet strike. This is useful for simulating the “chiff“ sound of a felt-wrapped mallet head. The Volume and Noise parameters can be modulated by pitch and velocity by adjusting the K (Key) and V (Velocity) sliders, respectively.

The Stiffness knob adjusts the hardness of the mallet. At low levels, the mallet is soft, which results in fewer high frequencies and a longer, less distinct impact. As you increase the stiffness, the impact time decreases and high frequencies increase. This parameter can also be modulated by pitch and velocity via the Key and Vel sliders.

The Color knob sets the frequency of the noise component. At higher values, there are less low frequencies in the noise. This parameter has no effect if Noise is set to 0.

The Mallet section can be toggled on or off via the switch next to its name.

The Noise Section

Collision’s Noise Section.

The Noise section can be used instead of, or in addition to, the Mallet section. Like the Mallet, the Noise section produces Collision’s initial impulse sound. But Noise also produces a white noise component, which is then fed into a multimode filter and a dedicated envelope generator.

Volume controls the overall output level of the Noise section, and can be modulated by pitch and velocity by adjusting the K (Key) and V (Velocity) sliders below the knob, respectively.

To the right are the filter controls. The type chooser allows you to choose between low-pass, high-pass, and two types of band-pass filters. Filter cutoff and resonance can be adjusted by the sliders above the filter display, or by dragging within the display itself. In BP mode, the second slider adjusts resonance, while in LP+HP mode, it adjusts bandwidth. The filter frequency can also be modulated by note pitch, velocity or the envelope generator, via the K, V and E sliders below the display.

The attack time — how quickly Noise reaches full volume — is set with the A (Attack) slider, while the time it takes for the envelope to reach the sustain level after the attack phase is set by the D (Decay) slider.

The S (Sustain) slider sets the level at which the envelope will remain from the end of the decay phase to the release of the key. When this slider is set to 0, there is no sustain phase. With it set to 100, there is no decay phase.

Finally, the release time is set with the R (Release) slider. This is the time it takes for the envelope to reach zero after the key is released.

The Noise section can be toggled on or off via the switch next to its name.

24.2.3
Resonator Tabs

Collision’s Resonators.

The majority of Collision’s character is determined by the parameters in the two Resonator tabs. Each stereo resonator can be toggled on or off via the switch in its tab. Keep in mind that if both resonators are turned off, no sound will be produced.

Each resonator section is further divided into three subsections. On the left are controls related to tuning. In the center are the controls that determine the physical properties of the resonant object. To the right are mixer controls. Each resonator’s center subsection contains a “Copy to“ button. Pressing this button copies this subsection’s parameter settings from the currently selected resonator to the other resonator.

The “link“ tab between the Resonator tabs allows you to adjust parameters for both resonators simultaneously. The behavior here is similar to what happens when you edit the properties for a multi-selection of clips (see Chapter 8): parameters that have differing values will display the value
range
(either on the control for knobs or in the status bar for sliders and choosers) and can be adjusted with the linked control. Dragging the parameter to its absolute maximum or minimum value will make the settings thereafter identical, adjustable as a single value.

Tuning Section

Resonator Tuning Parameters.

The Tune and Fine knobs function as coarse and fine tuning controls. Tune moves up or down in semitone increments, while Fine adjusts in increments of one cent (up to a maximum of one quarter tone (50 cents) up or down).

The Key slider below the Tune knob controls how much the resonator’s tuning is adjusted by changes in MIDI note pitch. The default value of 100% means that the resonator will conform to a conventional equal tempered scale. At 200%, each half step on the keyboard will result in a whole step change in tuning. At negative values, the resonator will drop in pitch as you play higher on the keyboard.

The Pitch Envelope parameters apply a ramp that modulates the resonator’s pitch over time. Pitch sets the starting pitch while Time adjusts how long it will take the pitch to glide to its final value. The starting pitch can be modulated by velocity via the Vel slider.

Physical Properties Section

Physical Properties of the Resonator.

The type chooser allows you to select from seven types of physically modeled resonant objects:

Beam
simulates the resonance properties of beams of different materials and sizes.

Marimba
, a specialized variant of the Beam model, reproduces the characteristic tuning of marimba bar overtones which are produced as a result of the deep arch-cut of the bars.

String
simulates the sound produced by strings of different materials and sizes.

Membrane
is a model of a rectangular membrane (such as a drum head) with a variable size and construction.

Plate
simulates sound production by a rectangular plate (a flat surface) of different materials and sizes.

Pipe
simulates a cylindrical tube that is fully open at one end and has a variable opening at the other (adjusted with the Opening parameter.)

Tube
simulates a cylindrical tube that is closed at both ends.

The quality chooser controls the tradeoff between the sound quality of the resonators and performance by reducing the number of overtones that are calculated. “Basic“ uses minimal CPU resources, while “Full“ creates more sophisticated resonances. This parameter is not used with the Pipe or Tube resonators.

The Decay slider adjusts the internal damping of the resonator, which determines its decay time. Off Decay determines the extent to which MIDI note off messages mute the resonance. At 0%, note offs are ignored, and the decay time is based only on the value of the Decay parameter. This is similar to how real-world mallet instruments such as a marimbas and glockenspiels behave. At 100%, the resonance is muted immediately at note off, regardless of the Decay time.

The Material slider adjusts the variation of the damping at different frequencies. At lower values, low frequency components decay slower than high frequency components (which simulates objects made of wood, nylon or rubber). At higher values, high frequency components decay slower (which simulates objects made of glass or metal). This parameter is not used with the Pipe or Tube resonators.

The Radius parameter is only available for the Pipe and Tube resonators. This slider adjusts the radius of the pipe or tube. As the radius increases, the decay time and high frequency sustain both increase. At very large sizes, the fundamental pitch of the resonator also changes.

The Decay and Material/Radius parameters can also be controlled with the X-Y controller, and modulated by note pitch and velocity via the sliders below the X-Y panel.

Ratio is only available for the Membrane and Plate resonators, and adjusts the ratio of the object’s size along its x and y axes.

The Brightness control adjusts the amplitude of various frequency components. At higher values, higher frequencies are louder. This parameter is not used with the Pipe or Tube resonators.

The Inharmonics knob adjusts the pitch of the resonator’s harmonics. At negative values, frequencies are compressed, increasing the amount of lower partials. At positive values, frequencies are stretched, increasing the amount of upper partials. This parameter is not used with the Pipe or Tube resonators. Inharmonics can also be modulated by velocity via the slider below the knob.

Opening, which is only available for the Pipe resonator, scales between an open and closed pipe. At 0%, the pipe is fully closed on one side, while at 100% the pipe is open at both ends. This parameter can also be modulated by velocity via the slider below the knob.

The Listening L and R controls adjust the location on the left and right resonator where the vibrations are measured. At 0%, the resonance is monitored at the object’s center. Higher values move the listening point closer to the edge. These parameters are not used with the Pipe or Tube resonators, which are always measured in the middle of their permanently open end.

The Hit knob adjusts the location on the resonator at which the object is struck or otherwise activated. At 0%, the object is hit at its center. Higher values move the activation point closer to the edge. This parameter is not used with the Pipe or Tube resonators. The Hit position can also be randomized by increasing the value of the Rd. (Random) slider below the knob.

Mixer Section

Resonator Mixer.

Each resonator has its own Volume and Pan controls. Pan can also be modulated by note pitch via the K (Key) slider below the knob.

The Bleed control mixes a portion of the original oscillator signal with the resonated signal. At higher values, more of the original signal is applied. This is useful for restoring high frequencies, which can often be damped when the tuning or quality are set to low values.

24.2.4
LFO Tab

Collision’s LFOs.

Collision’s two independent LFOs can be used as modulation sources for a variety of excitator and resonator parameters, which are selectable in the Destination choosers. Additionally, they can modulate each other.

The LFO 1 and LFO 2 switches toggle the respective LFO on and off, while the waveform chooser determines the wave shape. The choices are sine, square, triangle, sawtooth up, sawtooth down and two types of noise. The first noise type steps between random values while the second uses smooth ramps.

The switch next to the waveform chooser toggles the LFO’s rate between frequency in Hertz and tempo-synced beat divisions.

Depth sets the overall intensity of the LFO, while Rate adjusts its speed. The sliders below these parameters allow for additional modulations; Depth can be modulated by velocity while Rate can be modulated by note pitch.

With Retrig. enabled, triggering a note restarts the LFO with the waveform phase set by the Offset parameter. The Offset slider adjusts the phase.

Each LFO can modulate two targets, which are set via the Destination choosers. The intensity of the modulations is adjusted with the Amount sliders. Note that these modulation amounts are relative to the LFO’s Depth value.

24.2.5
MIDI Tab

Collision’s MIDI Tab.

The MIDI tab allows for a wide variety of internal MIDI mappings. The MIDI controllers
Pitch Bend, Modulation Whee
l, and
Aftertouch
can be mapped to two destinations each, with independent modulation intensities set via the Amount sliders. Note that pitch bend is hardwired to pitch modulation, but can still be routed to an additional target.

The Global Section

Collision’s Global Section.

The global section contains the parameters that relate to the overall behavior and performance of Collision.

The Volume knob acts as Collision’s master output control.

Collision contains a built-in limiter that automatically activates when the audio level is too high. This is indicated by the LED above Collision’s global Volume control.

The Structure buttons determine whether Collision’s resonators are arranged in
series
(1 > 2) or in
parallel
(1 + 2).

When in series, Mallet and Noise output to Resonator 1. This resonator’s output is then mixed down to mono and routed to Resonator 2, as well as to its own mixer (in stereo.) Note that Resonator 1 must be turned on when using serial mode:

Resonators in 1 > 2 (Serial) Configuration.

In parallel mode, the output of Mallet and Noise is mixed and then sent directly to both resonators, which then output to their own mixers.

Resonators in 1 + 2 (Parallel) Configuration.

The Voices chooser sets the available polyphony. Since each voice that’s used requires additional CPU, you may need to experiment with this chooser to find a good balance between playability and performance, particularly on older machines.

With Retrig. enabled, notes which are already sounding will be immediately stopped when retriggered, rather than generating an additional voice. This can be useful for keeping CPU down when working with long decay times.

24.2.6
Sound Design Tips

Although Collision has been designed to model the behavior of objects that exist in the physical world, it is important to remember that these models allow for much more flexibility than their physical counterparts. While Collision can produce extremely realistic simulations of conventional mallet instruments such as marimbas, vibraphones and glockenspiels, it is also very easy to “misuse“ the instrument’s parameters to produce sounds which could never be made by an acoustic instrument.

To program realistic instrument simulations, it helps to think about the chain of events that produces a sound on a mallet instrument (a marimba, for example), and then visualize those events as sections within Collision:

a beater (Mallet) strikes a tuned bar (Resonator 1).

the tuned bar’s resonance is amplified by means of a resonating tube (Resonator 2).

Thus the conventional model consists of the Mallet excitator and the two resonators in a serial (1 > 2) configuration.

Of course, to program
unrealistic
sounds, anything goes:

try using the Noise excitator, particularly with long envelope times, to create washy, quasi-granular textures. These parameters can also be used to simulate special acoustic effects such as bowed vibraphones or crystal glasses.

experiment with the resonators in parallel (1 + 2) configuration.

use the LFOs and MIDI controllers to modulate Collision’s parameters.

A word of caution
: in many ways, Collision’s models are idealized versions of real-world objects. Consequently, it is very easy to program resonances that are much more sensitive to input than any physical resonator could be. Certain combinations of parameters can cause dramatic changes in volume. Make sure to keep output levels low when experimenting with new sounds.

24.3
Electric

The Electric Instrument.

Electric is a software electric piano based on the classic instruments of the seventies, and developed in collaboration with Applied Acoustics Systems. Each component of these instruments has been modeled using cutting edge physical modeling technology to provide realistic and lively sounds. Physical modeling uses the laws of physics to reproduce the behavior of an object. In other words, Electric solves, in real time, mathematical equations describing how its different components function. No sampling or wavetables are used in Electric; the sound is simply calculated in real time by the CPU according to the values of each parameter. Electric is more than a simple recreation of vintage instruments; its parameters can be tweaked to values not possible with the real instruments to get some truly amazing new sounds that still retain a warm acoustic quality.

The full version of Electric is not included with the standard version of Live, but is a special feature available for purchase separately.

24.3.1
Architecture and Interface

The mechanism of the electric piano is actually quite simple. A note played on the keyboard activates a
mallet
that hits a
fork
. The sound of that fork is then amplified by a magnetic coil
pickup
and sent to the output, very much like an electric guitar. The fork is made of two parts, called the
tine bar
and
tone bar
. The tine bar is where the mallet hits the fork while the tone bar is a tuned metal resonator, sized appropriately to produce the correct pitch. Once the fork is activated, it will continue to resonate on its own for a long time. But releasing the key applies a
damper
to the fork, which mutes it more quickly.

The Electric interface is divided into five main sections, some of which are further divided into related subsections. The first four main sections (
Mallet, Fork, Damper
and
Pickup
) correspond to the sound producing components mentioned above. The
Global
section contains parameters that affect overall behavior and performance, such as pitch bend and polyphony.

24.3.2
Mallet Section

The Mallet section contains the parameters related to the physical properties of the mallet itself, as well as how it’s affected by your playing.

The Stiffness control adjusts the hardness of the mallet’s striking area. Higher values simulate a harder surface, which results in a brighter sound. Lower values mean a softer surface and a more mellow sound. The Force knob adjusts the intensity of the mallet’s impact on the fork. Low values simulate a soft impact while high values mean a hard impact.

The stiffness and force can also be modified by velocity and note pitch, via the Vel and Key sliders found below the knobs.

The Noise subsection simulates the impact noise caused by the mallet striking the fork. The Decay knob adjusts how long it takes for this noise to fade to silence, while the Pitch control sets the center frequency. Level adjusts the overall volume of the noise component. An additional Key scaling control adjusts how much the noise volume is determined by note pitch.

24.3.3
Fork Section

The Fork section is further divided into Tine and Tone subsections. This area is the heart of Electric’s sound generating mechanism.

The Tine subsection controls the portion of the fork that is directly struck by the mallet. The Decay knob adjusts how long it takes the tine’s sound to fade out while a note is held. The Color knob controls the relative amplitude of high and low partials in the tine’s spectrum. Low values increase the amount of low harmonics, while higher values result in higher harmonics. The amplitude of the tine is adjusted with the Level knob. This level can be further modulated by note pitch via the Key scaling control.

The Tone subsection controls the secondary resonance of the fork. Decay and Level parameters here work in the same way as their Tine counterparts.

The Release knob applies to both the Tine and Tone areas and controls the decay time of the fork’s sound after a key is released.

24.3.4
Damper Section

The metal forks in an electric piano are designed to sustain for a long time when a key is held. The mechanism that regulates this sustain is called the damper. When a key is pressed, that note’s damper is moved away from its fork. When the key is released, the damper is applied to the fork again to stop it from vibrating. But the dampers themselves make a small amount of sound, both when they are applied and when they are released. This characteristic noise is modelled in Electric’s Damper section.

The Tone knob adjusts the stiffness of the dampers. Turning this control to the left simulates soft dampers, which produces a mellower sound. Turning it to the right increases the hardness of the dampers, producing a brighter sound. The overall amount of damper noise is adjusted with the Level control.

The Att/Rel knob adjusts whether or not damper noise is present when the dampers are applied to the fork or when they are released. When turned to the left, damper noise is only present during the attack phase of the note. When turned to the right, the noise is present only during the release phase. In the center, an equal amount of noise will be added during both the attack and release.

24.3.5
Pickup Section

The Pickup section simulates the behavior of the magnetic coil pickup that amplifies the sound of the resonating fork.

The R-W buttons switch between two different types of pickups. In the R position, Electric simulates electro-dynamic pickups, while W is based on an electro-static model.

The Input knob is used to adjust the amount of the fork’s signal that is fed to the pickup, which in turn affects the amount of distortion applied to the overall signal. The Output knob controls the amount of signal output by the pickup section. Different combinations of these two knobs can yield very different results. For example, a low amount of input with a high amount of output will produce a cleaner sound than a high input with a low output. The output level can be further modulated by note pitch via the Key scaling control.

The Symmetry and Distance knobs adjust the physical location of the pickup in relation to the tine. Symmetry simulates the vertical position of the pickup. In the center position, the pickup is directly in front of the tine, which results in a brighter sound. Turning the knob to the left or right moves the pickup below or above the tine, respectively. Distance controls how far the pickup is from the tine. Turning the knob to the right increases the distance, while turning it to the left moves the pickup closer. Note that the sound becomes more overdriven as the pickup approaches the tine.

24.3.6
Global Section

The Global section contains the parameters that relate to the overall behavior and performance of Electric.

The Volume knob sets Electric’s overall output level.

The Voices chooser sets the available polyphony. Since each voice that’s used requires additional CPU, you may need to experiment with this chooser to find a good balance between playability and performance, particularly on older machines.

The Semi and Detune controls function as coarse and fine tuners. Semi transposes the entire instrument up or down in semitone increments, while the Detune slider adjusts in increments of one cent (up to a maximum of 50 cents up or down).

Stretch simulates a technique known as stretch tuning, which is a common modification made to both electric and acoustic pianos and is an intrinsic part of their characteristic sound. At 0%, Electric will play in equal temperament, which means that two notes are an octave apart when the upper note’s fundamental pitch is exactly twice the lower note’s. But because the actual resonance behavior of a vibrating tine or string differs from the theoretical model, equal temperament tends to sound “wrong“ on pianos. Stretch tuning attempts to correct this by sharpening the pitch of upper notes while flattening the pitch of lower ones. The result is a more brilliant sound. Negative values simulate “negative“ stretch tuning; upper notes become flatter while lower notes become sharper.

P Bend sets the range in semitones of pitch bend modulation.

24.4
External Instrument

(Note: the External Instrument device is not available in the Intro and Lite Editions.)

The External Instrument.

The External Instrument device is not an instrument itself, but rather a routing utility that allows you to easily integrate external (hardware) synthesizers, ReWire devices and multitimbral plug-ins into your projects. It sends MIDI out and returns audio.

The two MIDI To choosers select the output to which the device will send MIDI data. The top chooser selects either a physical MIDI port (see 14.3.1), a ReWire slave destination (see 14.4) or a multitimbral plug-in. If you select a MIDI port (for use with an external synthesizer), the second chooser’s options will be MIDI channel numbers. If you’ve chosen a ReWire slave such as Reason as your routing target, the choices will be the specific devices available in the slave project:

ReWire Options Shown in the Routing Choosers.

If another track in your set contains a multitimbral plug-in, you can select this track in the top chooser. In this case, the second chooser allows you to select a specific MIDI channel in the plug-in.

The Audio From chooser provides options for returning the audio from the hardware synth, plug-in, or ReWire device. If you’re routing to a hardware synth, use this chooser to select the ports on your audio interface that are connected to the output of your synth. The available choices you’ll have will depend on the settings in the Audio Preferences.

If you’re routing to a ReWire slave, the Audio From chooser will list all of the audio channels available in the slave. Select the audio channel that corresponds to the instrument to which you are sending MIDI. If you’re routing to a multitimbral plug-in on another track in your Live Set, the Audio From chooser will list the auxiliary outputs in the plug-in. Note that the main outputs will be heard on the track that contains the instrument.

The Gain knob adjusts the audio level coming back from the sound source. This level should be set carefully to avoid clipping.

Since external devices can introduce latency that Live cannot automatically detect, you can manually compensate for any delays by adjusting the Hardware Latency slider. The button next to this slider allows you to set your latency compensation amount in either milliseconds or samples. If your external device connects to Live via a digital connection, you will want to adjust your latency settings in samples, which ensures that the number of samples you specify will be retained even when changing the sample rate. If your external device connects to Live via an analog connection, you will want to adjust your latency settings in milliseconds, which ensures that the amount of time you specify will be retained when changing the sample rate. Note that adjusting in samples gives you finer control, so even in cases when you’re working with analog devices, you may want to “fine tune“ your latency in samples in order to achieve the lowest possible latency. In this case, be sure to switch back to milliseconds before changing your sample rate. Any latency introduced by devices within Live will be compensated for automatically, so the slider will be disabled when using the External Instrument Device to route internally. Latency adjustments when routing to ReWire devices will probably not be necessary, as most ReWire-enabled programs also compensate automatically. But if you feel that something is “off“ in the timing of your set, try adjusting this slider.

Note: If the Delay Compensation option (see 17.5) is unchecked in the Options menu, the Hardware Latency slider is disabled.

For more detailed information about routing scenarios with the External Instrument device, please see the Routing and I/O chapter (see Chapter 14).

24.5
Impulse

The Impulse Instrument.

Impulse is a drum sampler with complex modulation capabilities. The eight drum samples loaded into Impulse’s sample slots can be time-stretched, filtered and processed by envelope, saturation, pan and volume components, nearly all of which are subject to random and velocity-based modulation.

24.5.1
Sample Slots

Drag and drop samples into any of Impulse’s sample slots from the browser or the Session and Arrangement Views. Alternatively, each sample slot features a Hot-Swap button for hot-swapping samples (see “Hot-Swap Mode”). Loaded samples can be deleted with your computer keyboard’s
Backspace
or
Delete
key.

Imported samples are automatically mapped onto your MIDI keyboard, providing that it is plugged in and acknowledged by Live. C3 on the keyboard will trigger the leftmost sample, and the other samples will follow suit in the octave from C3 to C4. Impulse’s eight slots will appear labeled in the MIDI Editor’s key tracks (see Chapter 10) when the Fold button is active, even if the given key track is void of MIDI notes. Mapping can be transposed from the default by applying a Pitch device (see 23.4), or it can be rearranged by applying a Scale device (see 23.6).

Each of the eight samples has a proprietary set of parameters, located in the area below the sample slots and visible when the sample is clicked. Adjustments to sample settings are only captured once you hit a new note — they do not affect currently playing notes. Note that this behavior also defines how Impulse reacts to parameter changes from clip envelopes or automation, which are applied once a new note starts. If you want to achieve continuous changes as a note plays, you may want to use the Simpler (see 24.8).

Slot 8’s parameters also include a Link button, located in the lower left corner, which links slot 8 with slot 7. Linking the two slots allows slot 7’s activation to stop slot 8’s playback, and vice versa. This was designed with a specific situation in mind (but can, of course, be used for other purposes): Replicating the way that closed hi-hats will silence open hi-hats.

Each slot can be played, soloed, muted or hot-swapped using controls that appear when the mouse hovers over it.

24.5.2
Start, Transpose and Stretch

The Start control defines where Impulse begins playing a sample, and can be set up to 100 ms later than the actual sample beginning. The Transp (Transpose) control adjusts the transposition of the sample by +/- 48 semitones, and can be modulated by incoming note velocity or a random value, as set in the appropriate fields.

The Stretch control has values from -100 to 100 percent. Negative values will shorten the sample, and positive values will stretch it. Two different stretching algorithms are available: Mode A is ideal for low sounds, such as toms or bass, while Mode B is better for high sounds, such as cymbals. The Stretch value can also be modulated by MIDI note velocity.

24.5.3
Filter

The Filter section offers a broad range of filter types, each of which can impart different sonic characteristics onto the sample by removing certain frequencies. The Frequency control defines where in the harmonic spectrum the filter is applied; the Resonance control boosts frequencies near that point. Filter Frequency can be modulated by either a random value or by MIDI note velocity.

24.5.4
Saturator and Envelope

The Saturator gives the sample a fatter, rounder, more analog sound, and can be switched on and off as desired. The Drive control boosts the signal and adds distortion. Coincidentally, this makes most signals much louder, and should usually be compensated for by lowering the sample’s volume control. Extreme Drive settings on low-pitched sounds will produce the typical, overdriven analog synth drum sounds.

The envelope can be adjusted using the Decay control, which can be set to a maximum of 10.0 seconds. Impulse has two decay modes: Trigger Mode allows the sample to decay with the note; Gate Mode forces the envelope to wait for a note off message before beginning the decay. This mode is useful in situations where you need variable decay lengths, as is the case with hi-hat cymbal sounds.

24.5.5
Pan and Volume

Each sample has Volume and Pan controls that adjust amplitude and stereo positioning, respectively. Both controls can be modulated: Pan by velocity and a random value, and Volume by velocity only.

24.5.6
Global Controls

The parameters located to the right of the sample slots are global controls that apply to all samples within Impulse’s domain. Volume adjusts the overall level of the instrument, and Transp adjusts the transposition of all samples. The Time control governs the time-stretching and decay of all samples, allowing you to morph between short and stretched drum sounds.

24.5.7
Individual Outputs

When a new instance of Impulse is dragged into a track, its signal will be mixed with those of the other instruments and effects feeding the audio chain of the track. It can oftentimes make more sense to isolate the instrument or one of its individual drum samples, and send this signal to a separate track. Please see the Routing chapter (see “Tapping Individual Outs From an Instrument”) to learn how to accomplish this for Impulse’s overall signal or for Impulse’s individual sample slots.

24.6
Operator

The Operator Instrument.

Operator is an advanced and flexible synthesizer that combines the concept of “frequency modulation“ (FM) with classic subtractive and additive synthesis. It utilizes four multi-waveform oscillators that can modulate each other’s frequencies, creating very complex timbres from a limited number of objects. Operator includes a filter section, an LFO and global controls, as well as individual envelopes for the oscillators, filter, LFO and pitch.

With the release of Live 8, Operator has been dramatically overhauled with powerful new features. Despite this, the interface is mostly unchanged, and presets and Sets that were made with earlier versions of Operator are fully compatible with this update.

The full version of Operator is not included with the standard version of Live, but is a special feature available for purchase separately.

24.6.1
General Overview

The interface of Operator consists of two parts: the display surrounded on either side by the shell. The shell offers the most important parameters in a single view and is divided into eight sections. On the left side, you will find four oscillator sections, and on the right side from top to bottom, the LFO, the filter section, the pitch section and the global parameters. If you change one of the shell parameters, the display in the center will automatically show the details of the relevant section. When creating your own sounds, for example, you can conveniently access the level and frequency of all oscillators at once via the shell, and then adjust each individual oscillator’s envelope, waveform and other parameters in its display.

Operator can be folded with the triangular button at its upper left. This is convenient if you do not need to access the display details.

Operator Folded.

Each of Operator’s oscillators can either output its signal directly or use its signal to modulate another oscillator. Operator offers eleven predefined algorithms that determine how the oscillators are connected. An algorithm is chosen by clicking on one of the structure icons in the global display, which will appear if the bottom right (global) section of the shell is selected. Signals will flow from top to bottom between the oscillators shown in an algorithm icon. The algorithm selector can be mapped to a MIDI controller, automated, or modulated in real time, just like any other parameter.

Operator’s Global Display.

Typically, FM synthesis makes use of pure sine waves, creating more complex waveforms via modulation. However, in order to simplify sound design and to create a wider range of possible sounds, we designed Operator to produce a variety of other waveforms, including two types of noise. You can also draw your own waveforms via a partial editor. The instrument is made complete with an LFO, a pitch envelope and a filter section. Note that lots of “classic“ FM synthesizers create fantastic sounds without using filters at all, so we suggest exploring the possibilities of FM without the filter at first, and adding it later if necessary.

Operator will keep you busy if you want to dive deep into sound design! If you want to break the universe apart completely and reassemble it, you should also try modulating Operator’s controls with clip envelopes (see Chapter 20) or track automation (see Chapter 19).

24.6.2
Oscillator Section

Oscillator A’s Display and Shell Parameters.

Built-in Waveforms

The oscillators come with a built-in collection of basic waveform types — sine, sawtooth, square, triangle and noise — which are selected from the Wave chooser in the individual oscillator displays. The first of these waveforms is a pure, mathematical sine wave, which is usually the first choice for many FM timbres. We also added “Sine 4 Bit“ and “Sine 8 Bit“ to provide the retro sound adored by C64 fans, and “Saw D“ and “Square D“ digital waveforms, which are especially good for digital bass sounds. The square, triangle and sawtooth waveforms are resynthesized approximations of the ideal shape. The numbers included in the displayed name (e.g., “Square 6“) define how many harmonics are used for the resynthesis. Lower numbers sound mellower and are less likely to create aliasing when used on high pitches. There are also two built-in noise waveforms. The first, “Noise Looped,“ is a looping sample of noise. For truly random noise, choose “Noise White.“

User Waveforms

The “User“ entry in the Wave chooser allows you to create your own waveforms by drawing the amplitudes of the oscillator’s harmonics. You can also select one of the built-in waveforms and then edit it in the same way. The small display next to the Wave chooser gives a realtime overview of your waveform.

When your mouse is over the Oscillator display area, the cursor will change to a pencil. Drawing in the display area then raises or lowers the amplitudes of the harmonics. As you adjust the amplitudes, the Status Bar will show the number of the harmonic you’re adjusting as well as its amplitude. Holding
Shift
and dragging will constrain horizontal mouse movement, allowing you to adjust the amplitude of only one harmonic at a time.

You can switch between editing the first 16, 32 or 64 harmonics via the switches to the right of the display. Higher harmonics can be generated by repeating the drawn partials with a gradual fadeout, based on the settings in the Repeat chooser. Low Repeat values result in a brighter sound, while higher values result in more high-end roll-off and a more prominent fundamental. With Repeat off, partials above the 16th, 32nd or 64th harmonic are truncated.

The
right-click
(PC) /
CTRL
-
click
(Mac) context menu on the harmonics display offers options for editing only the even or odd harmonics. This is set to “All“ by default. The context menu also offers an option to toggle Normalize on or off. When enabled, the oscillator’s overall output level is maintained as you draw additional harmonics. When disabled, additional harmonics add additional level. Note that the volume can become extremely loud if Normalize is off.

You can export your waveform in
.ams
format to the Samples/Waveforms folder in your User Library via an option in the
right-click
(PC) /
CTRL
-
click
(Mac) context menu. Ams files can be imported back into Operator by dragging them from the browser onto one of the oscillator’s display areas. Ams files can also be loaded into Simpler or Sampler.

Hint: Both the built-in and user waveforms can be copied and pasted from one oscillator to another using the
right-click
(PC) /
CTRL
-
click
(Mac) context menu.

More Oscillator Parameters

The frequency of an oscillator can be adjusted in the shell with its Coarse and Fine controls. An oscillator’s frequency usually follows that of played notes, but for some sounds it might be useful to set one or more oscillators to fixed frequencies. This can be done for each individual oscillator by activating the Fixed option. This allows the creation of sounds in which only the timbre will vary when different notes are played, but the tuning will stay the same. Fixed Mode would be useful, for example, in creating live drum sounds. Fixed Mode also allows producing very low frequencies down to 0.1 Hz. Note that when Fixed Mode is active, the frequency of the oscillator is controlled in the shell with the Frequency (Freq) and Multiplier (Multi) controls.

Operator includes a special Osc

The amplitude of an oscillator depends on the Level setting of the oscillator in the shell and on its envelope, which is shown and edited when the Envelope display is visible. The envelopes can also be modified by note velocity and note pitch with the Vel and Key parameters available in the Envelope section of each oscillator’s display.

The phase of each oscillator can be adjusted using the Phase control in its display. With the R (Retrigger) button enabled, the waveform restarts at the same position in its phase each time a note is triggered. With R disabled, the oscillator is free-running.

As explained earlier oscillators can modulate each other when set up to do so with the global display’s algorithms. When an oscillator is modulating another oscillator, two main properties define the result: the amplitude of the modulating oscillator and the frequency ratio between both oscillators. Any oscillator that is not modulated by another oscillator can modulate itself, via the Feedback parameter in its display.

Aliasing

Aliasing distortion is a common side effect of all digital synthesis and is the result of the finite sample rate and precision of digital systems. It mostly occurs at high frequencies. FM synthesis is especially likely to produce this kind of effect, since one can easily create sounds with lots of high harmonics. This also means that more complex oscillator waveforms, such as “Saw 32,“ tend to be more sensitive to aliasing than pure sine waves. Aliasing is a two-fold beast: A bit of it can be exactly what is needed to create a cool sound, yet a bit too much can make the timbre unplayable, as the perception of pitch is lost when high notes suddenly fold back into arbitrary pitches. Operator minimizes aliasing by working in a high-quality Antialias mode. This is on by default for new patches, but can be turned off in the global section. The Tone parameter in the global section also allows for controlling aliasing. Its effect is sometimes similar to a lowpass filter, but this depends on the nature of the sound itself and cannot generally be predicted. If you want to familiarize yourself with the sound of aliasing, turn Tone up fully and play a few very high notes. You will most likely notice that some notes sound completely different from other notes. Now, turn Tone down and the effect will be reduced, but the sound will be less bright.

24.6.3
LFO Section

Operator’s LFO Parameters.

The LFO in Operator can practically be thought of as a fifth oscillator. It runs at audio rates, and it modulates the frequency of the other oscillators. It is possible to switch LFO modulation on or off for each individual oscillator (and the filter) using the Dest. A buttons in the LFO’s display. The intensity of the LFO’s modulation of these targets can be adjusted by the Dest. A slider. The LFO can also be turned off entirely if it is unused.

The Dest. B chooser allows the LFO to modulate an additional parameter. The intensity of this modulation is determined by the Dest. B slider.

The LFO offers a choice of classic LFO waveforms, sample and hold (S&H), and noise. Sample and hold uses random numbers chosen at the rate of the LFO, creating the random steps useful for typical retro-futuristic sci-fi sounds. The noise waveform is simply bandpass-filtered noise.

Tip: FM synthesis can be used to create fantastic percussion sounds, and using the LFO with the noise waveform is the key to great hi-hats and snares.

The frequency of the LFO is determined by the LFO Rate control in the shell, as well as the low/high/sync setting of the adjacent LFO Range chooser. The frequency of the LFO can follow note pitch, be fixed or be set to something in between. This is defined by the Rate

The overall intensity of the LFO is set by the LFO Amount control in the shell. This parameter scales both the Dest. A and B amounts and can be modulated by note velocity via the display’s Amt

24.6.4
Envelopes

Operator has seven envelopes: one for each oscillator, a filter envelope, a pitch envelope and an envelope for the LFO. All envelopes feature some special looping modes. Additionally, the filter and pitch envelopes have adjustable slopes.

Each oscillator’s volume envelope is defined by six parameters: three rates and three levels. A rate is the time it takes to go from one level to the next. For instance, a typical pad sound starts with the initial level “-inf dB“ (which is silence), moves with an attack rate to its peak level, moves from there to the sustain level with a decay rate, and then finally, after note-off occurs, back to “-inf dB“ at the release rate. Operator’s display provides a good overview of the actual shape of any particular envelope and lets you directly adjust the curve by clicking on a breakpoint and dragging. The breakpoints retain their selection after clicking, allowing them to be adjusted with the keyboard’s cursor keys, if desired.

Hint: Envelope shapes can be copied and pasted from one oscillator to another in Operator using the
right-click
(PC) /
CTRL
-
click
(Mac) context menu.

As mentioned above, the filter and pitch envelopes also have adjustable slopes. Clicking on the diamonds between the breakpoints allows you to adjust the slope of the envelope segments. Positive slope values cause the envelope to move quickly at the beginning, then slower. Negative slope values cause the envelope to remain flat for longer, then move faster at the end. A slope of zero is linear; the envelope will move at the same rate throughout the segment.

With FM synthesis, it is possible to create spectacular, endless, permuting sounds; the key to doing this is looping envelopes. Loop Mode can be activated in the lower left corner of the display. If an envelope in Operator is in Loop Mode and reaches sustain level while the note is still being held, it will be retriggered. The rate for this movement is defined by the Loop Time parameter. (Note that envelopes in Loop Mode can loop very quickly and can therefore be used to achieve effects that one would not normally expect from an envelope generator.)

While Loop Mode is good for textures and experimental sounds, Operator also includes Beat and Sync Modes, which provide a simple way of creating rhythmical sounds. If set to Beat Mode, an envelope will restart after the beat time selected from the Repeat chooser. In Beat Mode, the repeat time is defined in fractions of song time, but notes are not quantized. If you play a note a bit out of sync, it will repeat perfectly but stay out of sync. In Sync Mode however, the first repetition is quantized to the nearest 16th note and, as a result, all following repetitions are synced to the song tempo. Note that Sync Mode only works if the song is playing, and otherwise it will behave like Beat Mode.

Note: To avoid the audible clicks caused by restarting from its initial level, a looped envelope will restart from its actual level and move with the set attack rate to peak level.

There is also a mode called Trigger that is ideal for working with percussive sounds. In this mode, note off is ignored. This means that the length of time a key is held has no effect on the length of the sound.

The rates of all the envelopes in Operator can be scaled in unison by the Time control in the global section of the shell. Note that beat-time values in Beat and Sync Modes are not influenced by the global Time parameter. Envelope rates can be further modified by note pitch, as dictated by the Time

The pitch envelope can be turned on or off for each individual oscillator and for the LFO using the Destination A-D and LFO buttons in its display. The intensity of this envelope’s modulation of these targets can be adjusted by the Dest. A slider and the envelope can be turned off altogether via the switch in the pitch section of the shell.

Like the LFO, the pitch envelope can modulate an additional parameter as chosen by the Dest. B chooser. The intensity of this modulation is determined by the Amt. B slider and the main Pitch Env value.

The pitch and filter envelopes each have an additional parameter called End, which determines the level the envelope will move to after the key is released. The rate of this envelope segment is determined by the release time.

Tip: If the pitch envelope is only applied to the LFO and is looping, it can serve as another LFO, modulating the rate of the first. And, since the envelope of the LFO itself can loop, it can serve as a third LFO modulating the intensity of the first!

24.6.5
Filter Section

Operator’s Filter Section.

Operator’s filters can be very useful for modifying the sonically rich timbres created by the oscillators. And, since the oscillators also provide you with the classic waveforms of analog synthesizers, you can very easily build a subtractive synthesizer with them.

The filter section offers 14 different filter types including multiple varieties of lowpass, bandpass, highpass and notch filters. The 12 and 24 dB modes refer to the amount of attenuation. The 24 dB modes attenuate the filtered frequencies to a much greater degree than the 12 dB types, and are commonly used in the creation of bass patches. The SVF (state-variable filter) modes are 12 dB types but with a different architecture. They will self-oscillate as their resonance is increased. The Ladder modes have 24 dB slopes and are based on the filters found in some classic analog synthesizers.

The Envelope and Filter buttons in the filter section’s display area toggle between showing the filter envelope and its frequency response. Filter cutoff frequency and resonance can be adjusted in the shell or by dragging the filter response curve in the display area. Filter frequency can also be modulated by the following:

note velocity, via the Freq

note pitch, via the Freq

filter envelope, via the Envelope control in the filter’s display.

LFO, done either by enabling the Dest. A “FIL“ switch in the LFO’s display, or by setting Dest. B to Filter Freq.

The filter’s signal can be routed through a waveshaper, whose curve type can be selected via the Shaper chooser. The Drive slider boosts or attenuates the signal level being sent to the waveshaper, while the overall balance between the dry and processed signals can be adjusted with the Dry/Wet control. With this set to 0%, the shaper and drive parameters are bypassed.

24.6.6
Global Controls

The global section contains parameters that affect Operator’s overall behavior. Additionally, the global display area provides a comprehensive set of modulation routing controls.

The maximum number of Operator voices (notes) playing simultaneously can be adjusted with the Voices parameter in the global display. Ideally, one would want to leave this setting high enough so that no voices would be turned off while playing, however a setting between 6 and 12 is usually more realistic when considering CPU power.

Tip: Some sounds should play monophonically by nature, which means that they should only use a single voice. (A flute is a good example.) In these cases, you can set Voices to 1. If Voices is set to 1, another effect occurs: Overlapping voices will be played legato, which means that the envelopes will not be retriggered from voice to voice, and only pitch will change.

A global Volume control for the instrument can be found in the global section of the shell, and a Pan control is located in the global section’s display. Pan can be modulated by note pitch or a random factor, using the adjacent Pan

The center of the global display allows for a wide variety of internal MIDI mappings. The MIDI controllers Velocity, Key, Aftertouch, Pitch Bend and Mod Wheel can be mapped to two destinations each, with independent modulation intensities set via the Amount sliders. Note that Timesee 24.6.10).

24.6.7
Glide and Spread

Operator’s Pitch Section.

Operator includes a polyphonic glide function. When this function is activated, new notes will start with the pitch of the last note played and then slide gradually to their own played pitch. Glide can be turned on or off and adjusted with the Glide Time control in the pitch display.

Operator also offers a special Spread parameter that creates a rich stereo chorus by using two voices per note and panning one to the left and one to the right. The two voices are detuned, and the amount of detuning can be adjusted with the Spread control in the pitch section of the shell.

Tip: Whether or not spread is applied to a particular note depends upon the setting of the Spread parameter during the note-on event. To achieve special effects, you could, for instance, create a sequence where Spread is 0 most of the time and turned on only for some notes. These notes will then play in stereo, while the others will play mono. (Note: Spread is a CPU-intensive parameter.)

The pitch section also contains a global Transpose knob.

24.6.8
Strategies for Saving CPU Power

If you want to save CPU power, turn off features that you do not need or reduce the number of voices. Specifically, turning off the filter or the LFO if they do not contribute to the sound will save CPU power.

For the sake of saving CPU resources, you will also usually want to reduce the number of voices to something between 6 and 12, and carefully use the Spread feature. The Interpolation and Antialias modes in the global display can also be turned off to conserve CPU resources.

Note that turning off the oscillators will not save CPU power.

24.6.9
Finally...

Operator is the result of an intense preoccupation with FM synthesis and a love and dedication to the old hardware FM synthesizers, such as the Yamaha SY77, the Yamaha TX81Z and the NED Synclavier II. FM synthesis was first explored musically by the composer and computer music pioneer John Chowning in the mid-1960s. In 1973, he and Stanford University began a relationship with Yamaha that lead to one of the most successful commercial musical instruments ever, the DX7.

John Chowning realized some very amazing and beautiful musical pieces based on a synthesis concept that you can now explore yourself simply by playing with Operator in Live.

We wish you loads of fun with it!

24.6.10
The Complete Parameter List

The function of each Operator parameter is explained in the forthcoming sections. Remember that you can also access explanations of controls in Live (including those belonging to Operator) directly from the software by placing the mouse over the control and reading the text that appears in the Info View. Parameters in this list are grouped into sections based on where they appear in Operator.

Global Shell and Display

Time — This is a global control for all envelope rates.

Tone — Operator is capable of producing timbres with very high frequencies, which can sometimes lead to aliasing artifacts. The Tone setting controls the high frequency content of sounds. Higher settings are typically brighter but also more likely to produce aliasing.

Volume — This sets the overall volume of the instrument.

Algorithm — An oscillator can modulate other oscillators, be modulated by other oscillators, or both. The algorithm defines the connections between the oscillators and therefore has a significant impact on the sound that is created.

Voices — This sets the maximum number of notes that can sound simultaneously. If more notes than available voices are requested, the oldest notes will be cut off.

Retrigger (R) — When enabled, notes that are enabled will be retriggered, rather than generating an additional voice.

Interpolation — This toggles the interpolation algorithm of the oscillators and the LFO. If turned off, some timbres will sound more rough, especially the noise waveform. Turning this off will also save some CPU power.

Panorama — Modulates the position of Operator’s output in the stereo field.

Tone — Modulates the global Tone parameter.

Time — Modulates the global control for all envelope rates.

Pitch Shell and Display

Pitch Envelope On — This turns the pitch envelope on and off. Turning it off if it is unused saves some CPU power.

Pitch Envelope Amount (Pitch Env) — This sets the overall intensity of the pitch envelope. A value of 100% means that the pitch change is exactly defined by the pitch envelope’s levels. A value of -100% inverts the sign of the pitch envelope levels.

Spread — If Spread is turned up, the synthesizer uses two detuned voices per note, one each on the left and right stereo channels, to create chorusing sounds. Spread is a very CPU-intensive effect.

Transpose — This is the global transposition setting for the instrument. Changing this parameter will affect notes that are already playing.

Glide (G) — With Glide on, notes will slide from the pitch of the last played note to their played pitch. Note that all envelopes are not retriggered in this case if notes are being played legato.

Glide Time (Time) — This is the time it takes for a note to slide from the pitch of the last played note to its final pitch when Glide is activated. This setting has no effect if Glide is not activated.

Pitch Envelope to Osc (Destination A-D) — The pitch envelope affects the frequency of the respective oscillator if this is turned on.

Pitch Envelope to LFO (Destination LFO) — The pitch envelope affects the frequency of the LFO if this is turned on.

Pitch Envelope Amount A — This sets the intensity of the pitch envelope’s modulation of the oscillators and LFO.

Pitch Envelope Destination B — This sets the second modulation destination for the pitch envelope.

Pitch Envelope Amount B — This sets the intensity of the pitch envelope’s modulation of the secondary target.

Filter Shell and Display

Filter On — This turns the filter on and off. Turning it off when it is unused saves CPU power.

Filter Type — This chooser selects one of 14 filter types, including a variety of lowpass, highpass, bandpass and notch filters. The filter names imply the part of the spectrum they affect. A notch filter passes everything apart from its center frequency and is more audible with low resonance settings. The 24 dB filter modes attenuate the filtered frequencies to a much greater degree than the 12 dB modes. The Ladder and SVF filters provide additional filter architectures.

Filter Frequency (Freq) — This defines the center or cutoff frequency of the filter. Note that the resulting frequency may also be modulated by note velocity and by the filter envelope.

Filter Resonance (Res) — This defines the resonance around the filter frequency of the lowpass and highpass filters, and the width of the bandpass and notch filters.

Envelope / Filter Switches — These switches toggle the display between the filter’s envelope and its frequency response.

Filter Frequency

Filter Frequency

Filter Envelope Rates

Filter Frequency

Shaper — This chooser selects the curve for the filter’s waveshaper.

Drive — This boosts or attenuates the signal level being sent to the waveshaper.

Dry/Wet — This adjusts the balance between the dry signal and the signal processed by the waveshaper.

LFO Shell and Display

LFO On — This turns the LFO (low-frequency oscillator) on and off. Turning it off when it is unused saves some CPU power.

LFO Range — The LFO covers an extreme frequency range. Choose Low for a range from 50 seconds to 30 Hz, or Hi for 8 Hz to 12 kHz. Sync causes the LFO’s rate to be synced to your Set’s tempo. Due to the possible high frequencies, the LFO can also function as a fifth oscillator.

Retrigger (R) — When enabled, the LFO restarts at the same position in its phase each time a note is triggered. With R disabled, the LFO is free-running.

LFO Rate (Rate) — This sets the rate of the LFO. The actual frequency also depends on the setting of the LFO Range and the LFO Rate

LFO Amount (Amount) — This sets the overall intensity of the LFO. Note that the actual effect also depends on the LFO envelope.

LFO to Osc (Destination A-D) — The LFO modulates the frequency of the respective oscillator if this is turned on.

LFO to Filter Cutoff Frequency (Destination FIL) — The LFO modulates the cutoff frequency of the filter if this is turned on.

LFO Amount A — This sets the intensity of the LFO’s modulation of the oscillators and filter.

LFO Destination B — This sets the second modulation destination for the LFO.

LFO Amount B — This sets the intensity of the LFO’s modulation of the secondary target.

LFO Envelope Rates

LFO Rate

LFO Amount

Oscillators A-D Shell and Display

Osc On — This turns the oscillator on and off.

Osc Coarse Frequency (Coarse) — The relationship between oscillator frequency and note pitch is defined by the Coarse and Fine parameters. Coarse sets the ratio in whole numbers, creating a harmonic relationship.

Osc Fine Frequency (Fine) — The relationship between oscillator frequency and note pitch is defined by the Coarse and Fine parameters. Fine sets the ratio in fractions of whole numbers, creating an inharmonic relationship.

Osc Fixed Frequency On (Fixed) — In Fixed Mode, oscillators do not respond to note pitch but instead play a fixed frequency.

Osc Fixed Frequency (Freq) — This is the frequency of the oscillator in Hertz. This frequency is constant, regardless of note pitch.

Osc Fixed Multiplier (Multi) — This is used to adjust the range of the fixed frequency. Multiply this value with the value of the oscillator’s Freq knob to get actual frequency in Hz.

Osc Output Level (Level) — This sets the output level of the oscillator. If this oscillator is modulating another, its level has significant influence on the resulting timbre. Higher levels usually create bright and/or noisy sounds.

Envelope / Oscillator Switches — These switches toggle the display between the oscillator’s envelope and its harmonics editor.

16/32/64 — These switches set the number of partials that are available for user editing.

Osc Waveform (Wave) — Choose from a collection of carefully selected waveforms. You can then edit them via the harmonics editor.

Osc Feedback (Feedback) — An oscillator can modulate itself if it is not modulated by another oscillator. The modulation is dependent not only on the setting of the feedback control but also on the oscillator level and the envelope. Higher feedback creates a more complex resulting waveform.

Osc Phase (Phase) — This sets the initial phase of the oscillator. The range represents one whole cycle.

Retrigger (R) — When enabled, the oscillator restarts at the same position in its phase each time a note is triggered. With R disabled, the oscillator is free-running.

Repeat — Higher harmonics can be generated by repeating the drawn partials with a gradual fadeout, based on the settings in the Repeat chooser. Low Repeat values result in a brighter sound, while higher values result in more high-end roll-off and a more prominent fundamental. With Repeat off, partials above the 16th, 32nd or 64th harmonic are truncated.

Osc Frequency

Osc Freq

Volume Envelope Rates

Osc Output Level

Osc Output Level

Envelope Display

Envelope Attack Time (Attack) — This sets the time it takes for a note to reach the peak level, starting from the initial level. For the oscillator envelopes, the shape of this segment of the envelope is linear. For the filter and pitch envelopes, the shape of the segment can be adjusted.

Envelope Decay Time (Decay) — This sets the time it takes for a note to reach the sustain level from the peak level. For the oscillator envelopes, the shape of this segment of the envelope is exponential. For the filter and pitch envelopes, the shape of the segment can be adjusted.

Envelope Release Time (Release) — This is the time it takes for a note to reach the end level after a note-off message is received. For the oscillator envelopes, this level is always -inf dB and the shape of the segment is exponential. For the filter and pitch envelopes, the end level is determined by the End Level parameter and the shape of the segment can be adjusted. This envelope segment will begin at the value of the envelope at the moment the note-off message occurs, regardless of which segment is currently active.

Envelope Initial Level (Initial) — This sets the initial value of the envelope.

Envelope Peak Level (Peak) — This is the peak level at the end of the note attack.

Envelope Sustain Level (Sustain) — This is the sustain level at the end of the note decay. The envelope will stay at this level until note release unless it is in Loop, Sync or Beat Mode.

Envelope End Level (End) — (LFO, Filter and pitch envelopes only) This is the level reached at the end of the Release stage.

Envelope Loop Mode (Loop) — If this is set to Loop, the envelope will start again after the end of the decay segment. If set to Beat or Sync, it will start again after a given beat-time. In Sync Mode, this behavior will be quantized to song time. In Trigger mode, the envelope ignores note off.

Envelope Beat/Sync Rate (Repeat) — The envelope will be retriggered after this amount of beat-time, as long as it is still on. When retriggered, the envelope will move at the given attack rate from the current level to the peak level.

Envelope Loop Time (Time) — If a note is still on after the end of the decay/sustain segment, the envelope will start again from its initial value. The time it takes to move from the sustain level to the initial value is defined by this parameter.

Envelope Rates

The filter and pitch envelopes also provide parameters that adjust the slope of their envelope segments. Positive slope values cause the envelope to move quickly at the beginning, then slower. Negative slope values cause the envelope to remain flat for longer, then move faster at the end. A slope of zero is linear; the envelope will move at the same rate throughout the segment.

Harmonics editor commands — The context menu for the harmonics editor can restrict partial drawing to even or odd harmonics and toggle normalization of an oscillator’s output level. There is also a command to export the waveform as an
.ams
file.

Play By Key — This command, in the
right-click
(PC) /
CTRL
-
click
(Mac) context menu for the filter’s Freq control, optimizes the filter for key tracking by setting the cutoff to 466 Hz and setting the Freq

Live 8 Legacy Mode — This command, in the
right-click
(PC) /
CTRL
-
click
(Mac) context menu of Operator’s title bar, toggles the MIDI note that is the center point when using MIDI Key as a modulation source. When enabled, E3 is the center. When disabled, C3 is the center. Note that this option is only available when loading Operator presets that were made in versions of Live prior to Live 9.

24.7
Sampler

The Sampler Instrument.

Sampler is a sleek yet formidable multisampling instrument that takes full advantage of Live‘s agile audio engine. It has been designed from the start to handle multi-gigabyte instrument libraries with ease, and it imports most common library formats. But with Sampler, playback is only the beginning; its extensive internal modulation system, which addresses nearly every aspect of its sound, makes it the natural extension of Live‘s sound-shaping techniques.

The full version of Sampler is not included with the standard version of Live, but is a special feature available for purchase separately. Sampler users who want to share their presets with all Live users can convert their work to Simpler (see 24.8) presets. To do this,
right-click
(PC) /
CTRL
-
click
(Mac) on Sampler‘s title bar and choose the Sampler -> Simpler command.

24.7.1
Getting Started with Sampler

Getting started with Sampler is as easy as choosing a preset from the browser. Like all of Live‘s devices, Sampler’s presets are located in folders listed beneath its name. Presets imported from third-party sample libraries are listed here, too, in the
Imports
folder.

Once you have loaded a Sampler preset into a track, remember to arm the track for recording (which also enables you to hear any MIDI notes you might want to play), and then start playing!

24.7.2
Multisampling

Before going on, let’s introduce the concept of
multisampling
. This technique is used to accurately capture the complexity of instruments that produce dynamic timbral changes. Rather than rely on the simple transposition of a single recorded sample, multisampling captures an instrument at multiple points within its critical sonic range. This typically means capturing the instrument at different pitches as well as different levels of emphasis (played softly, moderately, loudly, etc.). The resulting
multisample
is a collection of all the individually recorded sample files.

The acoustic piano, for example, is a commonly multisampled instrument. Because the piano’s pitch and dynamic ranges are very wide and timbrally complex, transposing one sample across many octaves would not reproduce the nuances of the instrument. Since multisampling relies on different sound sources, three or more samples per piano key could be made (soft, medium, loud, very loud, and so on), maximizing the sampler’s expressive possibilities.

Sampler is designed to let you approach multisampling on whatever level you like: you can load and play multisample presets, import multisamples from third-party vendors (see 24.7.11), or create your own multisamples from scratch. Lastly, you do not have use multisamples at all — drop a single sample into Sampler and take advantage of its internal modulation system however you like.

24.7.3
Title Bar Options

Although Cut, Copy, Rename, Edit Info Text, and Delete should already be familiar, the other options deserve some explanation.

Group
— Selecting this will load Sampler into a new Instrument Rack.

Fold
— Folds Sampler so that only the device title bar is visible. Unfold quickly by double-clicking the device title bar.

Show Preset Name
— By default, Sampler takes the top-most sample in the sample layer list as its title. Unchecking Show Preset Name will replace the current title with “Sampler.”

Lock to Control Surface
— Locks Sampler to a natively supported control surface defined in the MIDI/Sync Preferences, guaranteeing hands-on access no matter where the current focus is in your Live Set. By default, Sampler will automatically be locked to the control surface when the track is armed for recording. A hand icon in the title bar of locked devices serves as a reminder of their statuses.

Save as Default Preset
— Saves the current state of Sampler as the default preset.

Use Constant Power Fade for Loops
— By default, Sampler uses constant-power fades at loop boundaries. Uncheck this to enable linear crossfades at looping points.

Sampler -> Simpler
— Converts Sampler presets to Simpler presets.

24.7.4
Sampler’s Tabs

Sampler’s features are organized categorically into tabs (Zone, Sample, Pitch/Osc, Filter/Global, Modulation and MIDI), accessed from Sampler‘s title bar. Clicking a tab will, with the exception of the Zone tab, reveal its properties below. In addition to serving as an organizational aid, each tab has one or more LEDs that indicate if there is modulation information in the corresponding area. We will get to know Sampler by examining each of these tabs.

Sampler’s Tabs in the Title Bar.

24.7.5
The Zone Tab

The Zone Tab.

Clicking on the Zone tab toggles the display of Sampler‘s Zone Editor, which offers a hands-on interface for mapping any number of samples across three types of ranges — the Key Zone, the Velocity Zone and Sample Select Editors, respectively.

The Key Zone Editor.

The Zone Editor opens in its own dedicated view, directly above the Device View. When used in conjunction with Sampler’s other tabs, this layout greatly accelerates the creation and editing of multisamples.

On the left side of the Zone Editor is the sample layer list, where multisamples are organized. All of the individual samples belonging to a multisample are shown in this list, where they are referred to as
layers
. For complex multisamples, this list can be quite long.

The rest of the view is occupied by one of three editors that correspond to the sample layers: the Key Zone Editor (see “Key Zones”), the Velocity Zone Editor (see “Velocity Zones”) and the Sample Select Editor (see “Sample Select Zones”). These editors can be horizontally zoomed by [right-clicking](PC) / [CTRL-clicking](Mac) within them to bring up a context menu, and selecting Small, Medium or Large.

Auto Select (Auto)
— As MIDI notes arrive at Sampler, they are filtered by each sample layer’s key, velocity and sample select zones. With Auto Select enabled, all sample layers that are able to play an incoming note will become selected in the sample layer list for the duration of that note.

Zone Fade Mode (Lin/Pow)
— These buttons toggle the fade mode of all zones between linear and constant-power (exponential) slopes.

The Sample Layer List

The Sample Layer List.

All samples contained in the currently loaded multisample are listed here, with each sample given its own layer. For very large multisamples, this list might be hundreds of layers long! Fortunately, layers can be descriptively named (according to their root key, for example). Mousing over a layer in the list or a zone in the zone editors will display relevant information about the corresponding sample in the Status Bar (bottom of screen). Selecting any layer will load its sample into the Sample tab for examination.

The Sample Layer List’s Context Menu

Pressing
right-click
(PC) /
CTRL
-
click
(Mac) within the sample layer list opens a context menu that offers options for sorting and displaying the layers, distributing them across the keyboard and various other sample management and “housekeeping“ options.

Distribute Ranges Around Root Key
— For layers that have different root keys, this option will distribute their ranges as evenly as possible around their root keys, but without overlapping. For layers that share a root key, the ranges will be distributed evenly.

Small/Medium/Large
— Adjusts the zoom level of the Zone Editor.

Show in Browser
— Navigates to the selected sample in the browser and selects it.

Manage Sample
— Opens the File Manager and selects the chosen sample.

Normalize Volume
— Adjusts Sampler’s Volume control so that the highest peak of each selected sample uses the maximum available headroom.

Normalize Pan
— Adjusts Sampler’s Pan control so that each selected sample has equal volume across the stereo spectrum. Note that this does not necessarily return panned stereo samples to the center position; rather, Live intelligently calculates a pan position for an even stereo spread.

Select All With Same Range
— Selects all layers whose zone range matches the currently selected layer. The results will change depending on which Zone Editor (Key, Velocity or Sample Select) is active.

Sort Alphabetically (Ascending and Descending)
— Arranges samples alphabetically according to their names.

Sort by Key (Ascending and Descending)
— Sorts key zones in an ascending or descending pattern.

Sort by Velocity (Ascending and Descending)
— Sorts velocity zones in an ascending or descending pattern.

Key Zones

The Key Zone Editor.

Key zones define the range of MIDI notes over which each sample will play. Samples are only triggered when incoming MIDI notes lie within their key zone. Every sample has its own key zone, which can span anywhere from a single key up to the full 127.

A typical multisampled instrument contains many individual samples, distributed into many key zones. Samples are captured at a particular key of an instrument’s voice range (known as their
root key
), but may continue to sound accurate when transposed a few semitones up or down. This range usually corresponds to the sample’s key zone; ranges beyond this zone are represented by additional samples, as needed.

By default, the key zones of newly imported samples cover the full MIDI note range. Zones can be moved and resized like clips in the Arrangement View, by dragging their right or left edges to resize them, then dragging them into position.

Zones can also be faded over a number of semitones at either end by dragging their top right or left corners. This makes it easy to smoothly crossfade between adjacent samples as the length of the keyboard is traversed. The Lin and Pow boxes above the sample layer list indicate whether the zones will fade in a linear or exponential manner.

Velocity Zones

The Velocity Zone Editor.

Velocity zones determine the range of MIDI Note On velocities (1-127) that each sample will respond to. The timbre of most musical instruments changes greatly with playing intensity. Therefore, the best multisamples capture not only individual notes, but also each of those notes at different velocities.

The Velocity Zone Editor, when toggled, appears alongside the sample layer list. Velocity is measured on a scale of 1-127, and this number range appears across the top of the editor. The functionality of the Velocity Zone Editor is otherwise identical to that of the Key Zone Editor.

Sample Select Zones

The Sample Select Editor.

Each sample also has a Sample Select zone, which is a data filter that is not tied to any particular kind of MIDI input. Sample Select zones are very similar to the Chain Select Zones (see 18.5.4) found in Racks, in that only samples with sample select values that overlap the current value of the
sample selector
will be triggered.

The Sample Select Editor, when toggled, appears alongside the sample layer list. The editor has a scale of 0-127, similar to the Velocity Zone Editor. Above the value scale is the draggable indicator known as the sample selector.

The Sample Selector.

Please note that the position of the sample selector only determines which samples are available for triggering. Once a sample has been triggered, changing the position of the sample selector will not switch to a different sample during playback.

24.7.6
The Sample Tab

The Sample Tab.

The playback behavior of individual samples is set within the Sample tab. Most of this tab is dedicated to displaying the waveform of the currently selected sample. Hovering your mouse over the waveform will display relevant information about the sample in the Status Bar (bottom of screen). It is important to keep in mind that most of the values in this tab reflect the state of the currently selected sample only. The Sample chooser always displays the current sample layer’s name, and is another way to switch between layers when editing.

Reverse
— This is a global, modulatable control that reverses playback of the entire multisample. Unlike the Reverse function in the Clip View, a new sample file is not generated. Instead, sample playback begins from the Sample End point, proceeds backwards through the Sustain Loop (if active), and arrives at the Sample Start point.

Snap
— Snaps all start and end points to the waveform zero-crossings (points where the amplitude is zero) to avoid clicks. You can quickly see this by using Snap on square wave samples. As with Simpler, this snap is based on the left channel of stereo samples, so a small Crossfade value may be necessary in some cases to completely eliminate clicks.

Sample
— Displays the name of the current sample layer, and can be used to quickly select different layers of the loaded multisample.

The Sample Chooser.

Root Key (RootKey)
— Defines the root key of the current sample.

Detune
— Sample tuning can be adjusted here by +/- 50 cents.

Volume
— A wide-range volume control, variable from full attenuation to a gain of +24 dB.

Pan
— Samples can be individually panned anywhere in the stereo panorama.

Sample Playback

All of the following parameters work in conjunction with the global volume envelope (in the Filter/Global tab) to create the basic voicing of Sampler. These envelopes use standard ADSR (Attack, Decay, Sustain, Release) parameters, among others:

Envelope Attack Time (Attack) — This sets the time it takes for an envelope to reach the peak level, starting from the initial level. The shape of the attack can be adjusted via the Attack Slope (A. Slope) parameter.

Envelope Decay Time (Decay) — This sets the time it takes for an envelope to reach the sustain level from the peak level. The shape of the decay can be adjusted via the Decay Slope (D. Slope) parameter.

Envelope Sustain Level (Sustain) — This is the sustain level at the end of the envelope decay. The envelope will stay at this level until note release unless it is in Loop, Sync or Beat Mode.

Envelope Release Time (Release) — This is the time it takes for an envelope to reach the end level after a note-off message is received. The shape of this stage of the envelope is determined by the Release Slope (R. Slope) value.

Envelope Initial Level (Initial) — This sets the initial value of the envelope.

Envelope Peak Level (Peak) — This is the peak level at the end of the envelope attack, and the beginning of the Decay stage.

Envelope End Level (End) — (LFO, Filter and pitch envelopes only) This is the level reached at the end of the Release stage.

Envelope Rates

Envelope Loop Mode (Loop) — If this is set to Loop, the envelope will start again after the end of the decay segment. If set to Beat or Sync, it will start again after a given beat-time. In Sync Mode, this behavior will be quantized to song time. In Trigger mode, the envelope ignores note off.

Envelope Beat/Sync Rate (Repeat) — The envelope will be retriggered after this amount of beat-time, as long as it is still on. When retriggered, the envelope will move at the given attack rate from the current level to the peak level.

Envelope Loop Time (Time) — If a note is still on after the end of the decay/sustain segment, the envelope will start again from its initial value. The time it takes to move from the sustain level to the initial value is defined by this parameter.

As mentioned above, Sampler’s envelopes also provide parameters that adjust the slope of their envelope segments. Positive slope values cause the envelope to move quickly at the beginning, then slower. Negative slope values cause the envelope to remain flat for longer, then move faster at the end. A slope of zero is linear; the envelope will move at the same rate throughout the segment.

All time-based values in this tab are displayed in either samples or minutes:seconds:milliseconds, which can be toggled using the
right-click
(PC) /
CTRL
-
click
(Mac) context menu on any of their parameter boxes.
Samples, in this context, refer to the smallest measurable unit in digital audio, and not to the audio files themselves, which we more commonly refer to as “samples.“

Sample Start
— The time value where playback will begin. If the volume envelope’s Attack parameter is set to a high value (slow attack), the audible result may begin some time later than the value shown here.

Sample End
— The time value where playback will end (unless a loop is enabled), even if the volume envelope has not ended.

Sustain Mode
— The optional Sustain Loop defines a region of the sample where playback will be repeated while the note is in the sustain stage of its envelope. Activating the Sustain Loop also allows the Release Loop to be enabled. This creates several playback options:

No Sustain Loop
— Playback proceeds linearly until either the Sample End is reached or the volume envelope completes its release stage.

Sustain Loop Enabled
— Playback proceeds linearly until Loop End is reached, when it jumps immediately to Loop Start and continues looping. If Release Mode is OFF, looping will continue inside the Sustain Loop until the volume envelope has completed its release stage.

Back-and-Forth Sustain Loop Enabled
— Playback proceeds to Loop End, then reverses until it reaches Loop Start, then proceeds again towards Loop End. If Release Mode is OFF, this pattern continues until the volume envelope has completed its release stage.

Link
— Enabling the Link switch sets Sample Start equal to Loop Start. Note that the Sample Start parameter box doesn’t lose its original value — it simply becomes disabled so that it can be recalled with a single click.

Loop Start
— The Sustain Loop’s start point, measured in samples.

Loop End
— The Sustain Loop’s end point, measured in samples.

Release Mode
— Whenever the Sustain Loop is active, Release Mode can also be enabled.

— The volume envelope’s release stage is active, but will occur within the Sustain Loop, with playback never proceeding beyond Loop End.

Release Loop Enabled
— When the volume envelope reaches its release stage, playback will proceed linearly until reaching Sample End, where it jumps immediately to Release Loop and continues looping until the volume envelope has completed its release stage.

Back-and-Forth Release Loop Enabled
— When the volume envelope reaches its release stage, playback will proceed linearly until reaching Sample End, then reverses until it reaches Release Loop, then proceeds again towards Sample End. This pattern continues until the volume envelope has completed its release stage.

Sustain and Release Loops.

Release Loop
— sets the start position of the Release Loop. The end of the Release Loop is the Sample End.

Sustain- and Release-Loop Detune (Detune)
— Since loops are nothing more than oscillations, the pitch of samples may shift within a loop, relative to the loop’s duration. Tip: this is especially noticeable with very short loops. With Detune, the pitch of these regions can be matched to the rest of the sample.

Sustain- and Release-Loop Detune Sliders.

Interpolation (Interpol)
— This is a global setting that determines the accuracy of transposed samples. Be aware that raising the quality level above “Normal“ to “Good” or “Best” will place significant demands on your CPU.

RAM Mode (RAM)
— This is also a global control that loads the entire multisample into RAM. This mode can give better performance when modulating start and end points, but loading large multisamples into RAM will quickly leave your computer short of RAM for other tasks. In any case, it is always recommended to have as much RAM in your computer as possible, as this can provide significant performance gains.

Hovering the mouse over the waveform and
right-click
(PC) /
CTRL
-
click
(Mac) provides a number of editing and viewing options. As with the context menu in the Sample Layer List, Show in Browser, Manage Samples, Normalize Volumes and Normalize Pan are available. Additionally, you can zoom in or out of playing or looping regions, depending on which Sustain and Loop Modes are selected.

Waveform Context Menu.

Finally, a few options remain on the far-right side of the Sample tab.

Vertical Zoom (slider)
— Magnifies the waveform height in the sample display. This is for visual clarity only, and does not affect the audio in any way.

B, M, L and R Buttons
— These buttons stand for Both, Mono, Left and Right, and allow you to choose which channels of the sample should be displayed.

The Sample Tab’s Vertical Zoom slider, and Channel Buttons.

24.7.7
The Pitch/Osc Tab

The Pitch/Osc Tab.

The Modulation Oscillator (Osc)

Sampler features one dedicated modulation oscillator per voice, which can perform frequency or amplitude modulation (
FM
or
AM
) on the multisample. The oscillator is fully featured, with 21 waveforms (available in the Type chooser), plus its own loopable amplitude envelope for dynamic waveshaping. Note that this oscillator performs modulation only — its output is never heard directly. What you hear is the effect of its output upon the multisample.

FM
— In this mode, the modulation oscillator will modulate the frequency of samples, resulting in more complex and different-sounding waveforms.

AM
— In this mode, the modulation oscillator will modulate the amplitude of samples. Subsonic modulator frequencies result in slow or rapid variation in the volume level; audible modulator frequencies result in composite waveforms.

The modulation oscillator is controlled via Initial, Peak, Sustain, End, Loop, Attack and Timesee “Sample Playback”). Additionally, the right side of the modulation oscillator section features the following controls:

Type
— Choose the modulation oscillator’s waveform here.

Volume
— This determines the intensity of the modulation oscillator’s sample modulation.

Vol
— The modulation oscillator’s Volume parameter can be modified by the velocity of incoming MIDI notes. This determines the depth of the modulation.

Fixed
— When enabled, the modulation oscillator’s frequency will remain fixed at the rate determined by the Freq and Multi parameters, and will not change in response to incoming MIDI notes.

Freq
— With Fixed set to On, this rate is multiplied by the Multi parameter to determine the modulation oscillator’s fixed frequency.

Multi
— With Fixed set to On, the Freq parameter is multiplied by this amount to determine the modulation oscillator’s fixed frequency.

Coarse
— Coarse tuning of the modulation oscillator’s frequency (0.125-48). This is only available when Fixed is set to Off.

Fine
— Fine tuning of the modulation oscillator’s frequency (0-1000). This is only available when Fixed is set to Off.

The Pitch Envelope

The pitch envelope modulates the pitch of the sample over time, as well as of the Modulation Oscillator, if it is enabled. This is a multi-stage envelope with ADSR, Initial, Peak, and End levels, as described in the Sample Playback section (see “Sample Playback”). The values of the envelope parameters can be adjusted via the sliders, or by dragging the breakpoints in the envelope’s display.

On the lower-left of the Pitch Envelope section is the Amount slider. This defines the limits of the pitch envelope’s influence, in semitones. The actual range depends upon the dynamics of the envelope itself.

The right-hand side of this section contains five sliders and one chooser that are unrelated to the Pitch Envelope, but can globally effect Sampler’s output:

The Lower Right-hand Corner of the Pitch/Osc Tab.

Spread
— When Spread is used, two detuned voices are generated per note. This also doubles the processing requirements.

Transpose (Transp)
— Global transpose amount, indicated in semitones.

Detune
— Global detune amount, indicated in cents.

Key Zone Shift (Zn Shft)
— This transposes MIDI notes in the Key Zone Editor only, so that different samples may be selected for playback, even though they will adhere to the played pitch. Good for getting interesting artifacts from multisamples.

Glide
— The global Glide mode, used in conjunction with the Time parameter to smoothly transition between pitches. ’Glide’ is a standard monophonic glide, while ’Portamento’ works polyphonically.

Time
— Enabling a Glide mode produces a smooth transition between the pitch of played notes. This parameter determines the length of the transition.

24.7.8
The Filter/Global Tab

The Filter/Shaper Order Button.

The Filter/Global Tab.

The Filter

Sampler features a polyphonic filter with an optional integrated waveshaper. The
Morph
(M12 and M24) and
SVF
filter types can morph continuously from lowpass to bandpass to highpass to notch and back to lowpass. Naturally, filter morphs can be automated.

Classic 24 dB lowpass, bandpass and highpass modes are also available, but these cannot be morphed.

To the right, the filter’s cutoff frequency can be modulated over time by a dedicated filter envelope. This envelope works similarly to the envelopes in the Pitch/Osc tab, with Initial, Peak, Sustain and End levels, ADSR, Loop mode and slope points. This area is toggled on/off with the F. Env button. The Amount slider determines how much influence the filter envelope has on the filter’s cutoff frequency, and needs to be set to a non-zero value for the envelope to have any effect.

Below the Filter is a waveshaper, which is toggled by clicking the Shaper button. Four different curves can be chosen for the waveshaper in the Type selector: Soft, Hard, Sine and 4bit. Shaper’s overall intensity can be controlled with the Amount slider. In addition, the signal flow direction can be adjusted with the button above the waveshaper area: with the triangle pointing up, the signal passes from the shaper to the filter; with the triangle pointing down, it passes from the filter to the shaper.

The Filter/Shaper Order Button.

The Volume Envelope

The Global Volume Envelope.

The volume envelope is global, and defines the articulation of Sampler’s sounds with standard ADSR (attack, decay, sustain, release) parameters. Please see the Sample Playback section (see “Sample Playback”) for details on these parameters.

This envelope can also be looped via the Loop chooser. When a Loop mode is selected, the Time/Repeat slider becomes important. For Loop and Trigger modes, if a note is still held when the Decay stage ends, the envelope will restart from its initial value. The time it takes to move from the Sustain level to the initial value is defined by the Time parameter. For Beat and Sync modes, if a note is still held after the amount set in the Repeat slider, the envelope will restart from its initial value.

The Pan slider is a global pan control (acting on all samples), while Pan
Controls For Global Pan and Global Time.

Finally, the Voices selector provides up to 32 simultaneous voices for each instance of Sampler. Voice retriggering can optionally be enabled by activating the Retrigger button (R) to the right of the Voices chooser. When activated, notes which are already playing will be retriggered, rather than generating an additional voice. Turning Retrigger on can save CPU power, especially if a note with a long release time is being triggered very often and very quickly.

24.7.9
The Modulation Tab

The Modulation Tab.

The Modulation tab offers an additional loopable envelope, plus three LFOs, all capable of modulating multiple parameters, including themselves. Each LFO can be free running, or synced to the Live Set’s tempo, and LFOs 2 and 3 can produce stereo modulation effects.

The Auxiliary Envelope

On the left, the Auxiliary (Aux) envelope functions much like the envelopes in the Pitch/Osc tab, with Initial, Peak, Sustain and End levels, ADSR, Loop mode and slope points. This envelope can be routed to 29 destinations in both the A and B choosers. How much the Auxiliary envelope will modulate destinations A and B is set in the two sliders to the right.

LFOs 1, 2 and 3

The remaining space of the Modulation tab contains three Low Frequency Oscillators (LFOs). As the name implies, Sampler’s LFOs operate by applying a low-frequency (below 30 Hz) to a parameter in order to modulate it. Engage any of these oscillators by clicking the LFO 1, LFO 2 or LFO 3 switches.

Rate
— With Hz selected, the speed of the LFO is determined by the Freq slider to the right. With the note head selected, the LFO will be synced to beat-time, adjustable in the Beats slider to the right.

Freq
— The LFO’s rate in Hertz (cycles per second), adjustable from 0.01 to 30 Hz.

Beats
— This sets the LFO’s rate in beat-time (64th notes to 8 bars).

LFO Attack (Attack)
— This is the time needed for the LFO to reach maximum intensity. Use this, for example, to gradually introduce vibrato as a note is held.

LFO Retrigger (Retrig)
— Enabling Retrigger for an LFO will cause it to reset to its starting point, or initial phase, on each new MIDI note. This can create hybrid LFO shapes if the LFO is retriggered before completing a cycle.

LFO Offset (Offset)
— This changes the starting point, or initial phase of an LFO, so that it begins at a different point in its cycle. This can create hybrid LFO shapes if the LFO is retriggered before completing a cycle.

Volume (Vol)
— LFO 1 can modulate the global volume level. This slider determines the depth of the modulation on a 0-100 scale.

Pan (Pan)
— LFO 1 can modulate the global pan position. This slider determines the depth of the modulation on a 0-100 scale.

Filter
— LFO 1 can modulate the filters cutoff frequency (Freq in the Filter/Global tab). This slider determines the depth of the modulation on a 0-24 scale.

Pitch
— LFO 1 can modulate the pitch of samples. This slider determines the depth of the modulation on a 0-100 scale.

LFO 1.

LFO Stereo Mode (Stereo)
— LFOs 2 and 3 can produce two types of stereo modulation:
Phase
or
Spin
. In phase mode, the right and left LFO channels run at equal speed, and the Phase parameter is used to offset the right channel from the left. In spin mode, the Spin parameter can make the right LFO channel run up to 50% faster than the left.

Like the Auxiliary envelope, LFOs 2 and 3 contain A and B choosers, where you can route LFOs to many destinations.

LFOs 2 and 3.

24.7.10
The MIDI Tab

The MIDI Tab.

The MIDI tab’s parameters turn Sampler into a dynamic performance instrument. The MIDI controllers
Key, Velocity, Release Velocity, Aftertouch, Modulation Wheel, Foot Controller
and
Pitch Bend
can be mapped to two destinations each, with varying degrees of influence determined in the Amount A and Amount B sliders.

For example, if we set Velocity’s Destination A to Loop Length, and its Amount A to 100, high velocities will result in long loop lengths, while low velocities will create shorter ones.

At the bottom is a Pitch Bend Range slider (0 to 24 steps). The 14-bit range of pitch wheel values can be scaled to produce up to 24 semitones of pitch bend in Sampler.

Finally, clicking in the Sampler image on the right will trigger a scrolling, movie-like credits for Sampler. These are the people you can thank!

24.7.11
Importing Third-Party Multisamples

Sampler can use multisamples created by a number of other software and hardware samplers. To import a third-party multisample, navigate to the file in Live‘s browser and drag it into a Live Set. This will import it into your User Library.

Importing will create new Sampler presets, which you can find in the browser under User Library/Sampler/Imports.

Note that some multisample files will be converted to Instrument Rack (see Chapter 18) presets that contain several Sampler instances used to emulate the original more accurately.

For all multisample formats except Apple EXS24/GarageBand and Kontakt, Live will import the actual audio data and create new samples. This means the new Sampler presets will work regardless of whether the original multisample file is still available.

To import Apple EXS24/GarageBand and Kontakt multisamples, Live will create new Sampler presets that reference the original WAV or AIF files. This means that removing the original WAV or AIF files will render the new Sampler presets useless. Live‘s File Manager offers the option to collect and save these external samples (see 5.8).

24.8
Simpler

The Simpler Instrument.

Simpler is an instrument that integrates the basic elements of a sampler with a set of classic synthesizer parameters. A Simpler voice plays a user-defined sample section, which is in turn processed by envelope, filter, LFO, volume and pitch components.

Presets created in Simpler can be converted for use in Sampler (see 24.7), and vice-versa. To do this,
right-click
(PC) /
CTRL
-
click
(Mac) on Simpler’s title bar and choose the Simpler -> Sampler command. In this way, presets created in Simpler can be further edited with Sampler’s extended functionality.

Owners of Sampler who wish to take their Simpler creations further can convert Simpler’s current settings to an identical setup in Sampler by
right-click
(PC) /
CTRL
-
click
(Mac) on Simpler’s title bar and choosing the Simpler -> Sampler command.

24.8.1
Sample View

The Sample View displays the sample waveform. Samples can be dragged into Simpler either directly from the browser, or from the Session or Arrangement View in the form of clips. In the latter case, Simpler will use only the section of the sample demarcated by the clip’s start/end or loop markers. Samples can be replaced by dragging in a new sample, or by activating the integrated Hot-Swap button.

24.8.2
Sample Controls

Simpler plays a specific region or loop of the sample, as determined by a group of sample controls.

The Start and Length controls work together to specify where Simpler begins and ends its sweep of the sample. As the name implies, Start defines where sample playback starts. The sample will play for the length defined by the Length parameter. Both parameters are defined as a percentage of the whole region, so setting start to 25 percent and length to 50 percent, for example, will start playback 1/4 of the way through the sample and stop playback at the 3/4 mark (using 50 percent of the sample).

Samples are played by Simpler as one-shots or as loops, depending on whether or not the Loop switch is active. When looping is active, the Loop control dictates the length of the loop, starting from the end of the playing sample. Simpler will play the first instance of a looped sample beginning with the Start point, then continue playing only the loop region.

When the sample’s start or end points are moved, Simpler will try to preserve the loop length for as long as possible by automatically adjusting the Start, Loop and Length settings.

It is possible for glitches or pops to occur between a looped sample’s start and end points due to the discontinuity in waveform amplitude (i.e., the sample’s loudness). The Snap switch will help mitigate these by forcing Simpler’s loop markers to snap to
zero-crossing
points in the sample (points where the amplitude is zero). Note: Snapping is based on the left channel of stereo samples. It is therefore still possible, even with Snap activated, to encounter glitches with stereo samples.

The transition from loop end to loop start can be smoothed with the Fade control, which crossfades the two points. This method is especially useful when working with long, textural samples. By default, Simpler uses constant-power fades. But by turning off “Use Constant Power Fade for Loops“ in the
right-click
(PC) /
CTRL
-
click
(Mac) context menu, you can enable linear crossfades.

24.8.3
Zoom

Quite often, one starts with a longer region of a sample and ends up using only a small part of it. Simpler’s Sample view can be zoomed and panned just as in other parts of Live — drag vertically to zoom, and drag horizontally to pan different areas of the sample into view.

24.8.4
Envelope

Simpler contains three classic ADSR envelopes, as seen in most synthesizers, for shaping the dynamic response of the sample. Volume-, filter frequency-, and pitch modulation are all modifiable by toggling their respective buttons in the envelope section.
Attack
controls the time in milliseconds that it takes for the envelope to reach its peak value after a note is played.
Decay
controls the amount of time it takes for the envelope to drop down to the
Sustain
level, which is held until the note is released.
Release
time is the amount of time after the end of the note that it takes for the envelope to drop from the Sustain level back down to zero.

The influence of envelopes on pitch and filter cutoff can be decided using the envelope amount (Env) controls in each of these sections.

24.8.5
Filter

The Filter section offers classic 12 dB or 24 dB lowpass, bandpass and highpass filters, as well as a notch filter, each of which can impart different sonic characteristics onto the sample by removing certain frequencies from the waveform. The most important parameters are the typical synth controls
Frequency
and
Resonance
. Frequency determines where in the harmonic spectrum the filter is applied; Resonance boosts frequencies near that point.

The best way to understand the effects of these controls is simply to play with them!

The Frequency parameter can be modulated by an LFO, note velocity and an envelope — each of which have an amount control in the Filter section. The Key (tracking) control allows for shifting the filter’s frequency according to note pitch.

24.8.6
LFO

The LFO (low-frequency oscillator) section offers sine, square, triangle, sawtooth down, sawtooth up and random waveforms. The LFO run freely at frequencies between 0.01 and 30 Hz, or sync to divisions of the Set’s tempo. LFOs are applied individually to each
voice
, or played note, in Simpler.

The Key parameter scales each LFO’s Rate in proportion to the pitch of incoming notes. A high Key setting assigns higher notes a higher LFO rate. If Key is set to zero, all voices’ LFOs have the same rate and may just differ in their phase.

The LFO will modulate the filter, pitch, panorama and volume according to the setting of the LFO amount controls in each of these sections.

The time required for the LFO to reach full intensity is determined by the Attack control.

24.8.7
Glide and Spread

Simpler includes a glide function. When this function is activated, new notes will start from the pitch of the last note played and then slide gradually to their own pitch. Two glide modes are available: Glide, which works monophonically, and Portamento, which works polyphonically. Glide is also adjusted with the Glide Time control.

Simpler also offers a special Spread parameter that creates a rich stereo chorus by using two voices per note and panning one to the left and one to the right. The two voices are detuned, and the amount of detuning can be adjusted with the Spread control.

Tip
: Whether or not spread is applied to a particular note depends upon the setting of the Spread parameter during the note-on event. To achieve special effects, you could, for instance, create a sequence where Spread is zero most of the time and turned on only for some notes. These notes will then play in stereo, while the others will play mono.

24.8.8
Pitch, Pan, Volume and Voices

Simpler plays back a sample at its original pitch if the incoming MIDI note is C3, however the Transpose control allows transposing this by +/- 48 semitones. Pitch can also be modulated by an LFO or pitch envelope using the amount controls in this section. The pitch envelope is especially helpful in creating percussive sounds. Simpler reacts to MIDI Pitch Bend messages with a sensitivity of +/- 5 semitones. You can also modulate the Transpose parameter with clip envelopes and external controllers.

The Voices parameter sets the maximum number of voices that Simpler can play simultaneously. If more voices are needed than have been allocated by the Voices chooser, “voice stealing“ will take place, in which the oldest voice(s) will be dropped in favor of those that are new. For example, if your Voices parameter is set to 8, and ten voices are all vying to be played, the two oldest voices will be dropped. (Simpler does try to make voice stealing as subtle as possible.)

Panorama is defined by the Pan control, but can be further swayed by randomness or modulated by the LFO.

Finally, the output volume of Simpler is controlled by the Volume control, which can also be dependent upon note velocity, as adjusted by the Velocity control. Tremolo effects can be achieved by allowing the LFO to modulate the Volume parameter.

24.8.9
Strategies for Saving CPU Power

Real-time synthesis needs lots of computing power. However, there are strategies for reducing CPU load. Save the CPU spent on Simpler by doing some of the following:

Turn off the Filter if it is not needed.

Use filter types that are less CPU-intensive when possible. A filter’s CPU cost correlates with the steepness of its slope — “LP24“ requires more CPU than “LP12.“

Turn off the LFO for a slightly positive influence on CPU.

Stereo samples need significantly more CPU than mono samples, as they require twice the processing.

Decrease the number of simultaneously allowed voices with the Voice control.

Turn Spread to 0% if it is not needed.

24.9
Tension

The Tension Instrument.

Tension is a synthesizer dedicated to the emulation of string instruments, and developed in collaboration with Applied Acoustics Systems. The synthesizer is entirely based on physical modeling technology and uses no sampling or wavetables. Instead, it produces sound by solving mathematical equations that model the different components in string instruments and how they interact. This elaborate synthesis engine responds dynamically to the control signals it receives while you play thereby reproducing the richness and responsiveness of real string instruments.

The full version of Tension is not included with the standard version of Live, but is a special feature available for purchase separately.

Tension features four types of excitators (two types of hammer, a pick and a bow) an accurate model of a string, a model of the fret/finger interaction, a damper model and different types of soundboards. The combination of these different elements allows for the reproduction of a wide range of string instruments. Tension is also equipped with filters, LFOs and envelope parameters that extend the sound sculpting possibilities beyond what would be possible with “real-world“ instruments. Finally, Tension offers a wide range of performance features, including keyboard modes, portamento, vibrato and legato functions.

24.9.1
Architecture and Interface

It is the vibration from the
string
which constitutes the main sound production mechanism of the instrument. The string is set into motion by the action of an
excitator
which can be a hammer, a pick or a bow. The frequency of the oscillation is determined by the effective length of the string, which is controlled by the finger/fret interaction or
termination
. A
damper
can be applied to the strings in order to reduce the decay time of the oscillation. This is the case on a piano, for example, when felt is applied to the strings by releasing the keys and sustain pedal. The vibration from the string is then transmitted to the
body
of the instrument, which can radiate sound efficiently. In some instruments, the string vibration is transmitted directly to the body through the bridge. In other instruments, such as the electric guitar, a
pickup
is used to transmit the string vibration to an amplifier. In addition to these main sections, a
filter
section has been included between the string and body sections in order to expand the sonic possibilities of the instrument.

The Tension interface is divided into two main tabs, which are further divided into sections. The
String
tab contains all of the fundamental sound producing components related to the string itself:
Excitator, String, Damper, Termination, Pickup
and
Body
. The
Filter/Global
tab contains the
Filter
section, as well as controls for global performance parameters. Each section (with the exception of String and the global Keyboard section) can be enabled or disabled independently. Turning off a section reduces CPU usage.

24.9.2
String Tab

The String tab contains the parameters related to the physical properties of the string itself, as well as the way in which it’s played.

The Excitator Section

Tension’s Excitator Section.

The modelled string can be played using different types of excitators in order to reproduce different types of instruments and playing techniques. The excitator is selected using the Type chooser, and the choices available are
Bow, Hammer, Hammer (bouncing)
and
Plectrum
.

Bow
— this excitator is associated with bowed instruments such as the violin, viola or cello. The bow sets the string in sustained oscillation. The motion of the bow hair across the string creates friction, causing the string to alternate between sticking to the hair and breaking free. The frequency of this alternation between sticking and slipping determines the fundamental pitch.

The Force knob adjusts the amount of pressure being applied to the string by the bow. The sound becomes more “scratchy“ as you increase this value. The friction between the bow and the string can be adjusted with the Friction control. Higher values usually result in a faster attack. Velocity adjusts the speed of the bow across the string. Finally, the Vel and Key sliders below these three controls allow you to modulate their behavior based on note velocity or pitch, respectively.

Hammer and Hammer (bouncing)
— these two excitator types simulate the behavior of soft hammers or mallets. Hammer models a hammer that is located below the string and strikes it once before falling away. This type of mechanism is found in a piano, for example. Hammer (bouncing) models a hammer that is located above the string and is dropped onto it, meaning that it can bounce on the string multiple times. This playing mode can be found on a hammered dulcimer, for example.

The mass and stiffness of the hammer are adjusted with the (surprise) Mass and Stiffness knobs, while Velocity controls the speed at which the hammer is struck against the string. As with the Bow excitator, these three parameters can be further modulated by note velocity and pitch by adjusting the Vel and Key sliders. The behavior of the hammer is further controlled by the Damping knob, which adjusts how much of the hammer’s impact force is absorbed back into the hammer. It is somewhat analogous to the Stiffness parameter, but instead of controlling the stiffness of the hammer’s surface it adjusts the stiffness of the virtual “spring“ that connects the hammer to the mass that powers it. As you increase the Damping amount, the interaction between the hammer and string will become shorter, generally resulting in a louder, brighter sound.

Plectrum
— a plectrum or “pick“ is associated with instruments such as guitars and harpsichords. It can be thought of as an angled object placed under the string that snaps the string into motion.

The Protrusion knob adjusts how much of the plectrum’s surface area is placed under the string. Lower values results in a “thinner,“ smaller sound, as there is less mass setting the string into motion. The Stiffness, Velocity and Damping knobs behave similarly to the Hammer mode. Protrusion, Stiffness, Velocity and Position can be modulated by velocity or note pitch via the Vel and Key sliders.

The Position knob is applicable to each excitator model, and specifies the point on the string where the excitator makes contact. At 0%, the excitator contacts the string at its termination point, while at 50% it activates the string at its midpoint. The behavior is a bit different if the Fix. Pos switch is enabled, however. In this case, the contact point is fixed to a single location, rather than changing as the length of the string changes. This behavior is similar to that of a guitar, where the picking position is always basically the same regardless of the notes being played. On a piano, the excitator position is relative — the hammers normally strike the string at about 1/7th of their length — and so is best modelled with Fix. Pos turned off. The excitator’s position can additionally be modulated by velocity or note pitch, via the Vel and Key sliders.

The Excitator section can be toggled on or off via the switch next to its name. With it off, the string can only be activated by interaction with its damper. (If both the Excitator and Damper sections are disabled, nothing can set the string in motion — if you find that you’re not producing any sound, check to see that at least one of these sections is on.)

Please note that the Excitator section’s parameters work closely together to influence the overall behavior of the instrument. You may find that certain combinations of settings result in no sound at all, for example.

The String Section

Tension’s String Section.

The vibration of the string is the main component of a stringed instrument’s sound. The effective length of the string is also responsible for the pitch of the sound we hear.

The theoretical model of a resonating string is harmonic, meaning that the string’s partials are all exact multiples of the fundamental frequency. Real-world strings, however, are all more or less inharmonic, and this increases with the width of the string. The Inharm slider models this behavior, causing upper partials to become increasingly out of tune as its value increases.

The Damping slider adjusts the amount of high frequency content in the string’s vibration. Higher values result in more upper partials (less damping). This parameter can be modulated by note pitch via the

The Decay slider determines how long it takes for the resonating string to decay to silence. Higher values increase the decay time. The

The Ratio slider sets the ratio of the decay time of the string’s oscillation during note onset and release. At 0%, the time set by the Decay slider sets the decay time for both the onset and release of the note. As you increase the Ratio, the release time decreases but the onset decay time stays the same.

The Vibrato Section

Tension’s Vibrato Section.

The Vibrato section uses an LFO to modulate the string’s pitch. As with all of Tension’s parameters, the controls in this section can be used to enhance the realism of a stringed instrument model — or to create something never heard before.

The two most important parameters in this section are the Rate and Amount sliders. Rate adjusts the frequency of the pitch variation, while Amount adjusts the intensity (amplitude) of the effect.

The Delay slider sets how long it will take for the vibrato to start after the note begins, while Attack sets how long it takes for the vibrato to reach full intensity (as set by the Amount knob).

The

The Error slider introduces unpredictability into the vibrato, by introducing random deviation to the Rate, Amount, Delay and Attack parameters.

The Damper Section

Tension’s Damper Section.

All string instruments employ some type of damping mechanism that mutes the resonating string. In pianos, this is a felt pad that is applied to the string when the key is released. In instruments such as guitars and violins, the player damps by stopping the string’s vibration with the fingers. Dampers regulate the decay of strings but also produce some sound of their own, which is an important characteristic of a string instrument’s timbre.

Although a damper functions to mute the string rather than activate it, it is somewhat analogous to a hammer, and shares some of the same parameters.

The Mass knob controls how hard the damper’s surface will press against the string. As you increase the value, the string will mute more quickly.

The stiffness of the damper’s material is adjusted with the Stiffness control. Lower values simulate soft materials such as felt, while higher values model a metal damper.

Note that very high Mass and Stiffness values can simulate dampers that connect with the string hard enough to change its effective length, thus causing a change in tuning.

The Velocity control adjusts the speed with which the damper is applied to the string when the key is released, as well as the speed with which it is lifted from the string when the key is depressed. Be careful with this parameter — very high Velocity values can cause the damper to hit the string extremely hard, which can result in a very loud sound on key release. Note that the state of the Gated switch determines whether or not the Velocity control is enabled. When the Gated switch is turned on, the damper is applied to the string when the key is released. With Gated off, the damper always remains on the string, which means that the Velocity control has no effect.

The Mass, Stiffness and Velocity parameters can be further modulated by note pitch, via the sliders below.

The stiffness of the damper mechanism is adjusted with the Damping knob, which affects the overall amount of vibration absorbed by the damper. Lower values result in less damping (longer decay times.) But this becomes a bit less predictable as the Damping value goes over 50%. At higher values, the mechanism becomes so stiff that it bounces against the string. This in turn reduces the overall amount of time that the damper is in contact with the string, causing an increase in decay time. The best way to get a sense of how this parameter behaves is to gradually turn up the knob as you repeatedly strike a single key.

The Position knob serves an analogous function to the control in the Excitator section, but here specifies the point on the string where the damper makes contact. At 0%, the damper contacts the string at its termination point, while at 50% it damps the string at its midpoint. The behavior is a bit different if the Fix. Pos switch is enabled, however. In this case, the contact point is fixed to a single location, rather than changing as the length of the string changes. The damper’s position can additionally be modulated by velocity or note pitch, via the Vel and Key sliders.

The Damper section can be toggled on or off via the switch next to its name.

The Termination Section

Tension’s Termination Section.

The Termination section models the interaction between the fret, finger and string. On a physical instrument, this interaction is used to change the effective length of the string, which in turn sets the pitch of the note played. The physical parameters of the finger are adjusted with the Fing Mass and Fing Stiff knobs, which set the force the finger applies to the string and the finger’s stiffness, respectively. The Mass amount can be additionally modulated by velocity or note pitch via the sliders. The stiffness of the fret is modelled with the Fret Stiff parameter.

The Pickup Section

Tension’s Pickup Section.

The Pickup section models an electromagnetic pickup, similar to the type found in an electric guitar or electric piano. The only control here is the Position slider, which functions similarly to this parameter in the Excitator and Damper sections. At 0%, the pickup is located at the string’s termination point, while at 50% it is under the midpoint of the string. Lower values generally result in a brighter, thinner sound, while higher values have more fullness and depth.

The Pickup section can be toggled on or off via the switch next to its name.

The Body Section

Tension’s Body Section.

The role of the body or soundboard of a string instrument is to radiate the vibration energy from the strings. The body also filters these vibrations, based on its size and shape. In some instruments, such as guitars, the body also includes an air cavity which boosts low frequencies.

The body type chooser allows you to select from different body types modelled after physical instruments.

The body size chooser sets the relative size of the resonant body, from extra small (XS) to extra large (XL). In general, as you increase the body size, the frequency of the resonance will become lower. You can further modify the body’s frequency response with the Hi Cut and Low Cut knobs.

The decay time of the body’s resonance can be adjusted with the Decay knob. Higher values mean a longer decay.

The Str/Body knob adjusts the ratio between the String section’s direct output and the signal filtered by the Body section. When turned all the way to the right, there is no direct output from the String section. When turned all the way to the left, the Body section is effectively bypassed.

The Body section can be toggled on or off via the switch next to its name.

The lonely Volume knob to the right of this section sets the overall output of the instrument. This knob is replicated on the Filter/Global tab.

24.9.3
Filter/Global Tab

The Filter/Global tab contains the filter parameters for the instrument, as well as global controls.

The Filter Section

Tension’s Filter Section.

Tension’s Filter section features a highly configurable multi-mode filter that sits between the String and Body sections. In addition, the filter can be modulated by a dedicated envelope generator and low-frequency oscillator (LFO).

The filter’s chooser allows you to select the filter type. You can choose between 2nd and 4th order low-pass, band-pass, notch, high-pass and formant filters.

The resonance frequency of the filter is adjusted with the Freq slider, while the amount of resonance is adjusted with the Res control. When a formant filter is chosen in the chooser, the Res control cycles between vowel sounds. The Freq and Res controls can each be modulated by LFO, envelope or note pitch via the sliders below. Note that the LFO and Env sliders have no effect unless the Envelope and LFO subsections are enabled.

The Envelope generator is a standard ADSR (attack, decay, sustain, release). This section can be toggled on or off via the switch next to its name.

The attack time is set with the Attack knob. This time can also be modulated by velocity via the Vel slider below the knob. As you increase the Vel value, the attack time will become increasingly shorter at higher velocities.

The time it takes for the envelope to reach the sustain level after the attack phase is set by the Decay knob.

The Sustain knob sets the level at which the envelope will remain from the end of the decay phase to the release of the key. When this knob is turned all the way to the left, there is no sustain phase. With it turned all the way to the right, there is no decay phase. The sustain level can be additionally modulated by velocity via the Vel slider below the knob. Higher values result in an increased sustain level as the velocity increases.

Finally, the release time is set with the Release knob. This is the time it takes for the envelope to reach zero after the key is released.

The LFO subsection provides an additional modulation source for the filter. This section can be toggled on or off via the switch next to its name.

The waveform chooser sets the type of waveform used by the LFO. You can choose between sine, triangle, rectangular and two types of random waveforms. The first random waveform steps between random values while the second uses smoother ramps.

The Delay knob sets how long it will take for the LFO to start after the note begins, while Attack sets how long it takes for the oscillator to reach its full amplitude.

The LFO’s speed is set with the Rate knob. The switches below this knob toggle the Rate between frequency in Hertz and tempo-synced beat divisions.

The entire Filter section can be toggled on or off via the switch next to its name.

Global and Keyboard Parameters

Tension’s Global and Keyboard Parameters.

The remaining section contain all of the parameters that adjust how Tension responds to MIDI data, as well as controls for performance parameters such as tuning and portamento.

The Keyboard section contains all of Tension’s polyphony and tuning parameters. The Voices chooser sets the available polyphony, while Priority determines which notes will be cut off when the maximum polyphony is exceeded. When Priority is set to High, new notes that are higher than currently sustained notes will have priority, and notes will be cut off starting from the lowest pitch. Low Priority is the opposite. A setting of Last gives priority to the most recently played notes, cutting off the oldest notes as necessary.

The Octave, Semi and Tuning controls function as coarse and fine tuners. Octave transposes the entire instrument by octaves, while Semi transposes up or down in semitone increments. The Tuning slider adjusts in increments of one cent (up to a maximum of 50 cents up or down).

The pitch bend modulation range in semitones is set by the P. Bend slider.

Stretch simulates a technique known as stretch tuning, which is a common tuning modification made to electric and acoustic pianos. At 0%, Tension will play in equal temperament, which means that two notes are an octave apart when the upper note’s fundamental pitch is exactly twice the lower note’s. But because the actual resonance behavior of a vibrating tine or string differs from the theoretical model, equal temperament tends to sound “wrong“ on pianos. Increasing the Stretch amount raises the pitch of upper notes while lowering the pitch of lower ones. The result is a more brilliant sound. Negative values simulate “negative“ stretch tuning; upper notes become flatter while lower notes become sharper.

The Error slider increases the amount of random tuning error applied to each note. Try very high values if you would like to relive your experiences from junior high school orchestra.

The Unison section allows you to stack multiple voices for each note played. The switch next to the name toggles the section on or off.

The Voices switch selects between two or four stacked voices, while Detune adjusts the amount of tuning variation applied to each stacked voice. Low values can create a subtle chorusing effect, while high values provide another good way to approximate a youth orchestra. Increasing the Delay amount adds lag before each stacked voice is activated.

The Portamento section is used to make the pitch slide between notes rather than changing immediately. The effect can be toggled on and off via the switch next to its name.

With Legato enabled, the sliding will only occur if the second note is played before the first note is released.

Proportional causes the slide time to be proportional to the interval between the notes. Large intervals will slide slower than small intervals. Disabling this switch causes the slide time to be constant regardless of interval.

The Time slider sets the overall speed of the slide.

The Volume knob sets the overall output of the instrument.

24.9.4
Sound Design Tips

At first glance, Tension’s modular architecture may not seem so different from what you’re used to in other synthesizers; it consists of functional building blocks that feed information through a signal path and modify it as it goes. But it’s important to remember that Tension’s components are not isolated from one another; what you do to one parameter can have a dramatic effect on a parameter somewhere else. Because of this, it’s very easy to find parameter combinations that produce no sound at all. It’s also very easy to create
extremely
loud sounds, so be careful when setting levels!

When programming Tension, it may help to think about the various sections as if they really are attached to a single, physical object. For example, a bow moving at a slow speed could perhaps excite an undamped string. But if that string is constricted by an enormous damper, the bow will need to increase its velocity to have any effect.

To get a sense of what’s possible, it may help to study how the presets were made. You’ll soon realize that Tension can do far more than just strings.